2 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
3 * <2006> Lutz Mueller <lutz at topfrose dot de>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
18 * Boston, MA 02111-1307, USA.
21 * Unless otherwise indicated, Source Code is licensed under MIT license.
22 * See further explanation attached in License Statement (distributed in the file
25 * Permission is hereby granted, free of charge, to any person obtaining a copy of
26 * this software and associated documentation files (the "Software"), to deal in
27 * the Software without restriction, including without limitation the rights to
28 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
29 * of the Software, and to permit persons to whom the Software is furnished to do
30 * so, subject to the following conditions:
32 * The above copyright notice and this permission notice shall be included in all
33 * copies or substantial portions of the Software.
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
37 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
38 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
44 * SECTION:element-rtspsrc
46 * Makes a connection to an RTSP server and read the data.
47 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
48 * RealMedia/Quicktime/Microsoft extensions.
50 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
51 * default rtspsrc will negotiate a connection in the following order:
52 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
53 * protocols can be controlled with the #GstRTSPSrc:protocols property.
55 * rtspsrc currently understands SDP as the format of the session description.
56 * For each stream listed in the SDP a new rtp_stream%d pad will be created
57 * with caps derived from the SDP media description. This is a caps of mime type
58 * "application/x-rtp" that can be connected to any available RTP depayloader
61 * rtspsrc will internally instantiate an RTP session manager element
62 * that will handle the RTCP messages to and from the server, jitter removal,
63 * packet reordering along with providing a clock for the pipeline.
64 * This feature is implemented using the gstrtpbin element.
66 * rtspsrc acts like a live source and will therefore only generate data in the
70 * <title>Example launch line</title>
72 * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
73 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
77 * Last reviewed on 2006-08-18 (0.10.5)
86 #endif /* HAVE_UNISTD_H */
93 #include <gst/sdp/gstsdpmessage.h>
94 #include <gst/rtp/gstrtppayloads.h>
96 #include "gst/gst-i18n-plugin.h"
98 #include "gstrtspsrc.h"
100 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
101 #define GST_CAT_DEFAULT (rtspsrc_debug)
103 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
106 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
108 /* templates used internally */
109 static GstStaticPadTemplate anysrctemplate =
110 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
113 GST_STATIC_CAPS_ANY);
115 static GstStaticPadTemplate anysinktemplate =
116 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
119 GST_STATIC_CAPS_ANY);
127 enum _GstRtspSrcRtcpSyncMode
134 enum _GstRtspSrcBufferMode
142 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
144 gst_rtsp_src_buffer_mode_get_type (void)
146 static GType buffer_mode_type = 0;
147 static const GEnumValue buffer_modes[] = {
148 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
149 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
150 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
151 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
155 if (!buffer_mode_type) {
157 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
159 return buffer_mode_type;
162 #define DEFAULT_LOCATION NULL
163 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
164 #define DEFAULT_DEBUG FALSE
165 #define DEFAULT_RETRY 20
166 #define DEFAULT_TIMEOUT 5000000
167 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
168 #define DEFAULT_TCP_TIMEOUT 20000000
169 #define DEFAULT_LATENCY_MS 2000
170 #define DEFAULT_CONNECTION_SPEED 0
171 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
172 #define DEFAULT_DO_RTCP TRUE
173 #define DEFAULT_PROXY NULL
174 #define DEFAULT_RTP_BLOCKSIZE 0
175 #define DEFAULT_USER_ID NULL
176 #define DEFAULT_USER_PW NULL
177 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
178 #define DEFAULT_PORT_RANGE NULL
179 #define DEFAULT_SHORT_HEADER FALSE
191 PROP_CONNECTION_SPEED,
200 PROP_UDP_BUFFER_SIZE,
205 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
207 gst_rtsp_nat_method_get_type (void)
209 static GType rtsp_nat_method_type = 0;
210 static const GEnumValue rtsp_nat_method[] = {
211 {GST_RTSP_NAT_NONE, "None", "none"},
212 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
216 if (!rtsp_nat_method_type) {
217 rtsp_nat_method_type =
218 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
220 return rtsp_nat_method_type;
223 static void gst_rtspsrc_finalize (GObject * object);
225 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
226 const GValue * value, GParamSpec * pspec);
227 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
228 GValue * value, GParamSpec * pspec);
230 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
231 gpointer iface_data);
233 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
236 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
237 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
239 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
241 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
242 GstStateChange transition);
243 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
244 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
246 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
247 GstRTSPMessage * response);
249 static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd);
250 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
251 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
253 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
254 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
256 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle,
258 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
259 gboolean only_close);
261 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
262 const gchar * uri, GError ** error);
264 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
265 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
266 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
267 GstRTSPStream * stream, GstEvent * event, gboolean source);
268 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event,
271 /* commands we send to out loop to notify it of events */
277 #define CMD_RECONNECT 5
280 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
282 gchar *__txt = _gst_element_error_printf text; \
283 gst_element_post_message (GST_ELEMENT_CAST (el), \
284 gst_message_new_progress (GST_OBJECT_CAST (el), \
285 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
289 /*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
290 #define gst_rtspsrc_parent_class parent_class
291 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
292 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
295 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
297 GObjectClass *gobject_class;
298 GstElementClass *gstelement_class;
299 GstBinClass *gstbin_class;
301 gobject_class = (GObjectClass *) klass;
302 gstelement_class = (GstElementClass *) klass;
303 gstbin_class = (GstBinClass *) klass;
305 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
307 gobject_class->set_property = gst_rtspsrc_set_property;
308 gobject_class->get_property = gst_rtspsrc_get_property;
310 gobject_class->finalize = gst_rtspsrc_finalize;
312 g_object_class_install_property (gobject_class, PROP_LOCATION,
313 g_param_spec_string ("location", "RTSP Location",
314 "Location of the RTSP url to read",
315 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
317 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
318 g_param_spec_flags ("protocols", "Protocols",
319 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
320 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
322 g_object_class_install_property (gobject_class, PROP_DEBUG,
323 g_param_spec_boolean ("debug", "Debug",
324 "Dump request and response messages to stdout",
325 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
327 g_object_class_install_property (gobject_class, PROP_RETRY,
328 g_param_spec_uint ("retry", "Retry",
329 "Max number of retries when allocating RTP ports.",
330 0, G_MAXUINT16, DEFAULT_RETRY,
331 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
333 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
334 g_param_spec_uint64 ("timeout", "Timeout",
335 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
336 0, G_MAXUINT64, DEFAULT_TIMEOUT,
337 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
339 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
340 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
341 "Fail after timeout microseconds on TCP connections (0 = disabled)",
342 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
343 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
345 g_object_class_install_property (gobject_class, PROP_LATENCY,
346 g_param_spec_uint ("latency", "Buffer latency in ms",
347 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
348 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
350 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
351 g_param_spec_uint64 ("connection-speed", "Connection Speed",
352 "Network connection speed in kbps (0 = unknown)",
353 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
354 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
356 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
357 g_param_spec_enum ("nat-method", "NAT Method",
358 "Method to use for traversing firewalls and NAT",
359 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
360 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
363 * GstRTSPSrc::do-rtcp
365 * Enable RTCP support. Some old server don't like RTCP and then this property
366 * needs to be set to FALSE.
370 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
371 g_param_spec_boolean ("do-rtcp", "Do RTCP",
372 "Send RTCP packets, disable for old incompatible server.",
373 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 * Set the proxy parameters. This has to be a string of the format
379 * [http://][user:passwd@]host[:port].
383 g_object_class_install_property (gobject_class, PROP_PROXY,
384 g_param_spec_string ("proxy", "Proxy",
385 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
386 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
389 * GstRTSPSrc::rtp_blocksize
391 * RTP package size to suggest to server.
395 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
396 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
397 "RTP package size to suggest to server (0 = disabled)",
398 0, 65536, DEFAULT_RTP_BLOCKSIZE,
399 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
401 g_object_class_install_property (gobject_class,
403 g_param_spec_string ("user-id", "user-id",
404 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
405 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
406 g_object_class_install_property (gobject_class, PROP_USER_PW,
407 g_param_spec_string ("user-pw", "user-pw",
408 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
409 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
412 * GstRTSPSrc::buffer-mode:
414 * Control the buffering and timestamping mode used by the jitterbuffer.
418 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
419 g_param_spec_enum ("buffer-mode", "Buffer Mode",
420 "Control the buffering algorithm in use",
421 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
422 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
425 * GstRTSPSrc::port-range:
427 * Configure the client port numbers that can be used to recieve RTP and
432 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
433 g_param_spec_string ("port-range", "Port range",
434 "Client port range that can be used to receive RTP and RTCP data, "
435 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
436 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
439 * GstRTSPSrc::udp-buffer-size:
441 * Size of the kernel UDP receive buffer in bytes.
445 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
446 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
447 "Size of the kernel UDP receive buffer in bytes, 0=default",
448 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
449 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
452 * GstRTSPSrc::short-header:
454 * Only send the basic RTSP headers for broken encoders.
458 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
459 g_param_spec_boolean ("short-header", "Short Header",
460 "Only send the basic RTSP headers for broken encoders",
461 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
463 gstelement_class->send_event = gst_rtspsrc_send_event;
464 gstelement_class->change_state = gst_rtspsrc_change_state;
466 gst_element_class_add_pad_template (gstelement_class,
467 gst_static_pad_template_get (&rtptemplate));
469 gst_element_class_set_details_simple (gstelement_class,
470 "RTSP packet receiver", "Source/Network",
471 "Receive data over the network via RTSP (RFC 2326)",
472 "Wim Taymans <wim@fluendo.com>, "
473 "Thijs Vermeir <thijs.vermeir@barco.com>, "
474 "Lutz Mueller <lutz@topfrose.de>");
476 gstbin_class->handle_message = gst_rtspsrc_handle_message;
478 gst_rtsp_ext_list_init ();
483 gst_rtspsrc_init (GstRTSPSrc * src)
485 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
486 src->protocols = DEFAULT_PROTOCOLS;
487 src->debug = DEFAULT_DEBUG;
488 src->retry = DEFAULT_RETRY;
489 src->udp_timeout = DEFAULT_TIMEOUT;
490 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
491 src->latency = DEFAULT_LATENCY_MS;
492 src->connection_speed = DEFAULT_CONNECTION_SPEED;
493 src->nat_method = DEFAULT_NAT_METHOD;
494 src->do_rtcp = DEFAULT_DO_RTCP;
495 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
496 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
497 src->user_id = g_strdup (DEFAULT_USER_ID);
498 src->user_pw = g_strdup (DEFAULT_USER_PW);
499 src->buffer_mode = DEFAULT_BUFFER_MODE;
500 src->client_port_range.min = 0;
501 src->client_port_range.max = 0;
502 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
503 src->short_header = DEFAULT_SHORT_HEADER;
505 /* get a list of all extensions */
506 src->extensions = gst_rtsp_ext_list_get ();
508 /* connect to send signal */
509 gst_rtsp_ext_list_connect (src->extensions, "send",
510 (GCallback) gst_rtspsrc_send_cb, src);
512 /* protects the streaming thread in interleaved mode or the polling
513 * thread in UDP mode. */
514 g_rec_mutex_init (&src->stream_rec_lock);
516 /* protects our state changes from multiple invocations */
517 g_rec_mutex_init (&src->state_rec_lock);
519 src->state = GST_RTSP_STATE_INVALID;
521 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
525 gst_rtspsrc_finalize (GObject * object)
529 rtspsrc = GST_RTSPSRC (object);
531 gst_rtsp_ext_list_free (rtspsrc->extensions);
532 g_free (rtspsrc->conninfo.location);
533 gst_rtsp_url_free (rtspsrc->conninfo.url);
534 g_free (rtspsrc->conninfo.url_str);
535 g_free (rtspsrc->user_id);
536 g_free (rtspsrc->user_pw);
539 gst_sdp_message_free (rtspsrc->sdp);
544 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
545 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
547 G_OBJECT_CLASS (parent_class)->finalize (object);
550 /* a proxy string of the format [user:passwd@]host[:port] */
552 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
556 g_free (rtsp->proxy_user);
557 rtsp->proxy_user = NULL;
558 g_free (rtsp->proxy_passwd);
559 rtsp->proxy_passwd = NULL;
560 g_free (rtsp->proxy_host);
561 rtsp->proxy_host = NULL;
562 rtsp->proxy_port = 0;
569 /* we allow http:// in front but ignore it */
570 if (g_str_has_prefix (p, "http://"))
573 at = strchr (p, '@');
575 /* look for user:passwd */
576 col = strchr (proxy, ':');
577 if (col == NULL || col > at)
580 rtsp->proxy_user = g_strndup (p, col - p);
582 rtsp->proxy_passwd = g_strndup (col, at - col);
587 col = strchr (p, ':');
590 /* everything before the colon is the hostname */
591 rtsp->proxy_host = g_strndup (p, col - p);
593 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
595 rtsp->proxy_host = g_strdup (p);
596 rtsp->proxy_port = 8080;
602 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
604 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
605 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
608 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
610 rtspsrc->ptcp_timeout = NULL;
614 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
619 rtspsrc = GST_RTSPSRC (object);
623 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
624 g_value_get_string (value), NULL);
627 rtspsrc->protocols = g_value_get_flags (value);
630 rtspsrc->debug = g_value_get_boolean (value);
633 rtspsrc->retry = g_value_get_uint (value);
636 rtspsrc->udp_timeout = g_value_get_uint64 (value);
638 case PROP_TCP_TIMEOUT:
639 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
642 rtspsrc->latency = g_value_get_uint (value);
644 case PROP_CONNECTION_SPEED:
645 rtspsrc->connection_speed = g_value_get_uint64 (value);
647 case PROP_NAT_METHOD:
648 rtspsrc->nat_method = g_value_get_enum (value);
651 rtspsrc->do_rtcp = g_value_get_boolean (value);
654 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
656 case PROP_RTP_BLOCKSIZE:
657 rtspsrc->rtp_blocksize = g_value_get_uint (value);
660 if (rtspsrc->user_id)
661 g_free (rtspsrc->user_id);
662 rtspsrc->user_id = g_value_dup_string (value);
665 if (rtspsrc->user_pw)
666 g_free (rtspsrc->user_pw);
667 rtspsrc->user_pw = g_value_dup_string (value);
669 case PROP_BUFFER_MODE:
670 rtspsrc->buffer_mode = g_value_get_enum (value);
672 case PROP_PORT_RANGE:
676 str = g_value_get_string (value);
678 sscanf (str, "%u-%u",
679 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
681 rtspsrc->client_port_range.min = 0;
682 rtspsrc->client_port_range.max = 0;
686 case PROP_UDP_BUFFER_SIZE:
687 rtspsrc->udp_buffer_size = g_value_get_int (value);
689 case PROP_SHORT_HEADER:
690 rtspsrc->short_header = g_value_get_boolean (value);
693 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
699 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
704 rtspsrc = GST_RTSPSRC (object);
708 g_value_set_string (value, rtspsrc->conninfo.location);
711 g_value_set_flags (value, rtspsrc->protocols);
714 g_value_set_boolean (value, rtspsrc->debug);
717 g_value_set_uint (value, rtspsrc->retry);
720 g_value_set_uint64 (value, rtspsrc->udp_timeout);
722 case PROP_TCP_TIMEOUT:
726 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
727 rtspsrc->tcp_timeout.tv_usec;
728 g_value_set_uint64 (value, timeout);
732 g_value_set_uint (value, rtspsrc->latency);
734 case PROP_CONNECTION_SPEED:
735 g_value_set_uint64 (value, rtspsrc->connection_speed);
737 case PROP_NAT_METHOD:
738 g_value_set_enum (value, rtspsrc->nat_method);
741 g_value_set_boolean (value, rtspsrc->do_rtcp);
747 if (rtspsrc->proxy_host) {
749 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
753 g_value_take_string (value, str);
756 case PROP_RTP_BLOCKSIZE:
757 g_value_set_uint (value, rtspsrc->rtp_blocksize);
760 g_value_set_string (value, rtspsrc->user_id);
763 g_value_set_string (value, rtspsrc->user_pw);
765 case PROP_BUFFER_MODE:
766 g_value_set_enum (value, rtspsrc->buffer_mode);
768 case PROP_PORT_RANGE:
772 if (rtspsrc->client_port_range.min != 0) {
773 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
774 rtspsrc->client_port_range.max);
778 g_value_take_string (value, str);
781 case PROP_UDP_BUFFER_SIZE:
782 g_value_set_int (value, rtspsrc->udp_buffer_size);
784 case PROP_SHORT_HEADER:
785 g_value_set_boolean (value, rtspsrc->short_header);
788 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
794 find_stream_by_id (GstRTSPStream * stream, gint * id)
796 if (stream->id == *id)
803 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
805 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
812 find_stream_by_pt (GstRTSPStream * stream, gint * pt)
814 if (stream->pt == *pt)
821 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
823 GstElement *src = (GstElement *) a;
825 if (stream->udpsrc[0] == src)
827 if (stream->udpsrc[1] == src)
834 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
836 /* check qualified setup_url */
837 if (!strcmp (stream->conninfo.location, (gchar *) a))
839 /* check original control_url */
840 if (!strcmp (stream->control_url, (gchar *) a))
843 /* check if qualified setup_url ends with string */
844 if (g_str_has_suffix (stream->control_url, (gchar *) a))
850 static GstRTSPStream *
851 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
855 /* find and get stream */
856 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
857 return (GstRTSPStream *) lstream->data;
862 static const GstSDPBandwidth *
863 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
864 const GstSDPMedia * media, const gchar * type)
868 /* first look in the media specific section */
869 len = gst_sdp_media_bandwidths_len (media);
870 for (i = 0; i < len; i++) {
871 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
873 if (strcmp (bw->bwtype, type) == 0)
876 /* then look in the message specific section */
877 len = gst_sdp_message_bandwidths_len (sdp);
878 for (i = 0; i < len; i++) {
879 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
881 if (strcmp (bw->bwtype, type) == 0)
888 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
889 const GstSDPMedia * media, GstRTSPStream * stream)
891 const GstSDPBandwidth *bw;
893 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
894 stream->as_bandwidth = bw->bandwidth;
896 stream->as_bandwidth = -1;
898 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
899 stream->rr_bandwidth = bw->bandwidth;
901 stream->rr_bandwidth = -1;
903 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
904 stream->rs_bandwidth = bw->bandwidth;
906 stream->rs_bandwidth = -1;
910 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
911 const GstSDPConnection * conn)
913 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
916 if (conn->addrtype == NULL)
920 if (strcmp (conn->addrtype, "IP4") == 0)
921 stream->is_ipv6 = FALSE;
922 else if (strcmp (conn->addrtype, "IP6") == 0)
923 stream->is_ipv6 = TRUE;
928 g_free (stream->destination);
929 stream->destination = g_strdup (conn->address);
931 /* check for multicast */
932 stream->is_multicast =
933 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
935 stream->ttl = conn->ttl;
938 /* Go over the connections for a stream.
939 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
941 * - If we are dealing with a localhost address, we disable multicast
944 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
945 const GstSDPMedia * media, GstRTSPStream * stream)
947 const GstSDPConnection *conn;
950 /* first look in the media specific section */
951 len = gst_sdp_media_connections_len (media);
952 for (i = 0; i < len; i++) {
953 conn = gst_sdp_media_get_connection (media, i);
955 gst_rtspsrc_do_stream_connection (src, stream, conn);
957 /* then look in the message specific section */
958 if ((conn = gst_sdp_message_get_connection (sdp))) {
959 gst_rtspsrc_do_stream_connection (src, stream, conn);
963 static GstRTSPStream *
964 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
966 GstRTSPStream *stream;
967 const gchar *control_url;
968 const gchar *payload;
969 const GstSDPMedia *media;
971 /* get media, should not return NULL */
972 media = gst_sdp_message_get_media (sdp, idx);
976 stream = g_new0 (GstRTSPStream, 1);
977 stream->parent = src;
978 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
980 stream->last_ret = GST_FLOW_NOT_LINKED;
981 stream->added = FALSE;
982 stream->disabled = FALSE;
983 stream->id = src->numstreams++;
985 stream->discont = TRUE;
986 stream->seqbase = -1;
987 stream->timebase = -1;
989 /* collect bandwidth information for this steam. FIXME, configure in the RTP
990 * session manager to scale RTCP. */
991 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
993 /* collect connection info */
994 gst_rtspsrc_collect_connections (src, sdp, media, stream);
996 /* we must have a payload. No payload means we cannot create caps */
997 /* FIXME, handle multiple formats. The problem here is that we just want to
998 * take the first available format that we can handle but in order to do that
999 * we need to scan for depayloader plugins. Scanning for payloader plugins is
1000 * also suboptimal because the user maybe just wants to save the raw stream
1001 * and then we don't care. */
1002 if ((payload = gst_sdp_media_get_format (media, 0))) {
1003 stream->pt = atoi (payload);
1005 stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
1007 GST_DEBUG ("mapping sdp session level attributes to caps");
1008 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
1009 GST_DEBUG ("mapping sdp media level attributes to caps");
1010 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
1012 if (stream->pt >= 96) {
1013 /* If we have a dynamic payload type, see if we have a stream with the
1014 * same payload number. If there is one, they are part of the same
1015 * container and we only need to add one pad. */
1016 if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
1017 stream->container = TRUE;
1018 GST_DEBUG ("found another stream with pt %d, marking as container",
1023 /* collect port number */
1024 stream->port = gst_sdp_media_get_port (media);
1026 /* get control url to construct the setup url. The setup url is used to
1027 * configure the transport of the stream and is used to identity the stream in
1028 * the RTP-Info header field returned from PLAY. */
1029 control_url = gst_sdp_media_get_attribute_val (media, "control");
1030 if (control_url == NULL)
1031 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1033 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1034 GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
1035 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1036 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1037 GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
1038 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1040 if (control_url != NULL) {
1041 stream->control_url = g_strdup (control_url);
1042 /* Build a fully qualified url using the content_base if any or by prefixing
1043 * the original request.
1044 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1045 * likely build a URL that the server will fail to understand, this is ok,
1046 * we will fail then. */
1047 if (g_str_has_prefix (control_url, "rtsp://"))
1048 stream->conninfo.location = g_strdup (control_url);
1053 if (g_strcmp0 (control_url, "*") == 0)
1057 base = src->control;
1058 else if (src->content_base)
1059 base = src->content_base;
1060 else if (src->conninfo.url_str)
1061 base = src->conninfo.url_str;
1065 /* check if the base ends or control starts with / */
1066 has_slash = g_str_has_prefix (control_url, "/");
1067 has_slash = has_slash || g_str_has_suffix (base, "/");
1069 /* concatenate the two strings, insert / when not present */
1070 stream->conninfo.location =
1071 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1074 GST_DEBUG_OBJECT (src, " setup: %s",
1075 GST_STR_NULL (stream->conninfo.location));
1077 /* we keep track of all streams */
1078 src->streams = g_list_append (src->streams, stream);
1086 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1090 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1093 gst_caps_unref (stream->caps);
1095 g_free (stream->destination);
1096 g_free (stream->control_url);
1097 g_free (stream->conninfo.location);
1099 for (i = 0; i < 2; i++) {
1100 if (stream->udpsrc[i]) {
1101 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1102 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1103 gst_object_unref (stream->udpsrc[i]);
1104 stream->udpsrc[i] = NULL;
1106 if (stream->channelpad[i]) {
1107 gst_object_unref (stream->channelpad[i]);
1108 stream->channelpad[i] = NULL;
1110 if (stream->udpsink[i]) {
1111 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1112 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1113 gst_object_unref (stream->udpsink[i]);
1114 stream->udpsink[i] = NULL;
1117 if (stream->fakesrc) {
1118 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1119 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1120 gst_object_unref (stream->fakesrc);
1121 stream->fakesrc = NULL;
1123 if (stream->srcpad) {
1124 gst_pad_set_active (stream->srcpad, FALSE);
1125 if (stream->added) {
1126 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1127 stream->added = FALSE;
1129 stream->srcpad = NULL;
1131 if (stream->rtcppad) {
1132 gst_object_unref (stream->rtcppad);
1133 stream->rtcppad = NULL;
1135 if (stream->session) {
1136 g_object_unref (stream->session);
1137 stream->session = NULL;
1143 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1147 GST_DEBUG_OBJECT (src, "cleanup");
1149 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1150 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1152 gst_rtspsrc_stream_free (src, stream);
1154 g_list_free (src->streams);
1155 src->streams = NULL;
1157 if (src->manager_sig_id) {
1158 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1159 src->manager_sig_id = 0;
1161 gst_element_set_state (src->manager, GST_STATE_NULL);
1162 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1163 src->manager = NULL;
1165 src->numstreams = 0;
1167 gst_structure_free (src->props);
1170 g_free (src->content_base);
1171 src->content_base = NULL;
1173 g_free (src->control);
1174 src->control = NULL;
1177 gst_rtsp_range_free (src->range);
1180 /* don't clear the SDP when it was used in the url */
1181 if (src->sdp && !src->from_sdp) {
1182 gst_sdp_message_free (src->sdp);
1187 #define PARSE_INT(p, del, res) \
1190 p = strstr (p, del); \
1200 #define PARSE_STRING(p, del, res) \
1203 p = strstr (p, del); \
1215 #define SKIP_SPACES(p) \
1216 while (*p && g_ascii_isspace (*p)) \
1221 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1224 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1225 gint * rate, gchar ** params)
1229 p = (gchar *) rtpmap;
1231 PARSE_INT (p, " ", *payload);
1239 PARSE_STRING (p, "/", *name);
1240 if (*name == NULL) {
1241 GST_DEBUG ("no rate, name %s", p);
1242 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1243 * streams seem to omit the rate. */
1250 p = strstr (p, "/");
1268 * Mapping SDP attributes to caps
1270 * prepend 'a-' to IANA registered sdp attributes names
1271 * (ie: not prefixed with 'x-') in order to avoid
1272 * collision with gstreamer standard caps properties names
1275 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1277 if (attributes->len > 0) {
1281 s = gst_caps_get_structure (caps, 0);
1283 for (i = 0; i < attributes->len; i++) {
1284 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1285 gchar *tofree, *key;
1289 /* skip some of the attribute we already handle */
1290 if (!strcmp (key, "fmtp"))
1292 if (!strcmp (key, "rtpmap"))
1294 if (!strcmp (key, "control"))
1296 if (!strcmp (key, "range"))
1299 /* string must be valid UTF8 */
1300 if (!g_utf8_validate (attr->value, -1, NULL))
1303 if (!g_str_has_prefix (key, "x-"))
1304 tofree = key = g_strdup_printf ("a-%s", key);
1308 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1309 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1316 * Mapping of caps to and from SDP fields:
1318 * m=<media> <UDP port> RTP/AVP <payload>
1319 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1320 * a=fmtp:<payload> <param>[=<value>];...
1323 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1326 const gchar *rtpmap;
1330 gchar *params = NULL;
1336 /* get and parse rtpmap */
1337 if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
1338 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1340 if (payload != pt) {
1341 /* we ignore the rtpmap if the payload type is different. */
1342 g_warning ("rtpmap of wrong payload type, ignoring");
1348 /* if we failed to parse the rtpmap for a dynamic payload type, we have an
1352 /* else we can ignore */
1353 g_warning ("error parsing rtpmap, ignoring");
1356 /* dynamic payloads need rtpmap or we fail */
1360 /* check if we have a rate, if not, we need to look up the rate from the
1361 * default rates based on the payload types. */
1363 const GstRTPPayloadInfo *info;
1365 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1366 /* dynamic types, use media and encoding_name */
1367 tmp = g_ascii_strdown (media->media, -1);
1368 info = gst_rtp_payload_info_for_name (tmp, name);
1371 /* static types, use payload type */
1372 info = gst_rtp_payload_info_for_pt (pt);
1376 if ((rate = info->clock_rate) == 0)
1379 /* we fail if we cannot find one */
1384 tmp = g_ascii_strdown (media->media, -1);
1385 caps = gst_caps_new_simple ("application/x-unknown",
1386 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1388 s = gst_caps_get_structure (caps, 0);
1390 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1392 /* encoding name must be upper case */
1394 tmp = g_ascii_strup (name, -1);
1395 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1399 /* params must be lower case */
1400 if (params != NULL) {
1401 tmp = g_ascii_strdown (params, -1);
1402 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1406 /* parse optional fmtp: field */
1407 if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
1413 /* p is now of the format <payload> <param>[=<value>];... */
1414 PARSE_INT (p, " ", payload);
1415 if (payload != -1 && payload == pt) {
1419 /* <param>[=<value>] are separated with ';' */
1420 pairs = g_strsplit (p, ";", 0);
1421 for (i = 0; pairs[i]; i++) {
1423 const gchar *val, *key;
1425 /* the key may not have a '=', the value can have other '='s */
1426 valpos = strstr (pairs[i], "=");
1428 /* we have a '=' and thus a value, remove the '=' with \0 */
1430 /* value is everything between '=' and ';'. We split the pairs at ;
1431 * boundaries so we can take the remainder of the value. Some servers
1432 * put spaces around the value which we strip off here. Alternatively
1433 * we could strip those spaces in the depayloaders should these spaces
1434 * actually carry any meaning in the future. */
1435 val = g_strstrip (valpos + 1);
1437 /* simple <param>;.. is translated into <param>=1;... */
1440 /* strip the key of spaces, convert key to lowercase but not the value. */
1441 key = g_strstrip (pairs[i]);
1442 if (strlen (key) > 1) {
1443 tmp = g_ascii_strdown (key, -1);
1444 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1456 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1461 g_warning ("rate unknown for payload type %d", pt);
1467 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
1468 gint * rtpport, gint * rtcpport)
1471 GstStateChangeReturn ret;
1472 GstElement *udpsrc0, *udpsrc1;
1473 gint tmp_rtp, tmp_rtcp;
1477 src = stream->parent;
1483 /* Start at next port */
1484 tmp_rtp = src->next_port_num;
1486 if (stream->is_ipv6)
1487 host = "udp://[::0]";
1489 host = "udp://0.0.0.0";
1491 /* try to allocate 2 UDP ports, the RTP port should be an even
1492 * number and the RTCP port should be the next (uneven) port */
1495 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
1496 tmp_rtp >= src->client_port_range.max)
1499 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1500 if (udpsrc0 == NULL)
1501 goto no_udp_protocol;
1502 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
1504 if (src->udp_buffer_size != 0)
1505 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
1508 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
1509 if (ret == GST_STATE_CHANGE_FAILURE) {
1511 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
1514 if (++count > src->retry)
1517 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1518 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1519 gst_object_unref (udpsrc0);
1521 GST_DEBUG_OBJECT (src, "retry %d", count);
1524 goto no_udp_protocol;
1527 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
1528 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
1530 /* check if port is even */
1531 if ((tmp_rtp & 0x01) != 0) {
1532 /* port not even, close and allocate another */
1533 if (++count > src->retry)
1536 GST_DEBUG_OBJECT (src, "RTP port not even");
1538 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1539 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1540 gst_object_unref (udpsrc0);
1542 GST_DEBUG_OBJECT (src, "retry %d", count);
1547 /* allocate port+1 for RTCP now */
1548 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1549 if (udpsrc1 == NULL)
1550 goto no_udp_rtcp_protocol;
1553 tmp_rtcp = tmp_rtp + 1;
1554 if (src->client_port_range.max > 0 && tmp_rtcp >= src->client_port_range.max)
1557 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
1559 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
1560 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
1561 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
1562 if (ret == GST_STATE_CHANGE_FAILURE) {
1563 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
1565 if (++count > src->retry)
1568 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1569 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1570 gst_object_unref (udpsrc0);
1572 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
1573 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1574 gst_object_unref (udpsrc1);
1578 GST_DEBUG_OBJECT (src, "retry %d", count);
1582 /* all fine, do port check */
1583 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
1584 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
1586 /* this should not happen... */
1587 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
1590 /* we keep these elements, we configure all in configure_transport when the
1591 * server told us to really use the UDP ports. */
1592 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
1593 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
1595 /* keep track of next available port number when we have a range
1597 if (src->next_port_num != 0)
1598 src->next_port_num = tmp_rtcp + 1;
1605 GST_DEBUG_OBJECT (src, "could not get UDP source");
1610 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
1614 no_udp_rtcp_protocol:
1616 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
1621 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
1622 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
1628 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1629 gst_object_unref (udpsrc0);
1632 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1633 gst_object_unref (udpsrc1);
1640 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
1647 GstClockTime base_time = GST_CLOCK_TIME_NONE;
1650 event = gst_event_new_flush_start ();
1651 GST_DEBUG_OBJECT (src, "start flush");
1653 state = GST_STATE_PAUSED;
1655 event = gst_event_new_flush_stop (TRUE);
1656 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
1659 state = GST_STATE_PLAYING;
1661 state = GST_STATE_PAUSED;
1662 clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
1664 base_time = gst_clock_get_time (clock);
1665 gst_object_unref (clock);
1668 gst_rtspsrc_push_event (src, event, FALSE);
1669 gst_rtspsrc_loop_send_cmd (src, cmd);
1671 /* set up manager before data-flow resumes */
1672 /* to manage jitterbuffer buffer mode */
1674 gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
1675 /* and to have base_time trickle further down,
1676 * e.g. to jitterbuffer for its timeout handling */
1677 if (base_time != -1)
1678 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
1681 /* make running time start start at 0 again */
1682 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1683 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1685 for (i = 0; i < 2; i++) {
1687 if (stream->udpsrc[i]) {
1688 if (base_time != -1)
1689 gst_element_set_base_time (stream->udpsrc[i], base_time);
1690 gst_element_set_state (stream->udpsrc[i], state);
1694 /* for tcp interleaved case */
1695 if (base_time != -1)
1696 gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
1699 static GstRTSPResult
1700 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
1701 GstRTSPMessage * message, GTimeVal * timeout)
1706 ret = gst_rtsp_connection_send (conn, message, timeout);
1708 ret = GST_RTSP_ERROR;
1713 static GstRTSPResult
1714 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
1715 GstRTSPMessage * message, GTimeVal * timeout)
1720 ret = gst_rtsp_connection_receive (conn, message, timeout);
1722 ret = GST_RTSP_ERROR;
1728 gst_rtspsrc_get_position (GstRTSPSrc * src)
1733 query = gst_query_new_position (GST_FORMAT_TIME);
1734 /* should be known somewhere down the stream (e.g. jitterbuffer) */
1735 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1736 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1740 if (stream->srcpad) {
1741 if (gst_pad_query (stream->srcpad, query)) {
1742 gst_query_parse_position (query, &fmt, &pos);
1743 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
1744 GST_TIME_ARGS (pos));
1745 src->last_pos = pos;
1755 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
1757 src->state = GST_RTSP_STATE_SEEKING;
1758 /* PLAY will add the range header now. */
1759 src->need_range = TRUE;
1765 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
1770 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
1772 gboolean flush, skip;
1775 GstSegment seeksegment = { 0, };
1779 GST_DEBUG_OBJECT (src, "doing seek with event");
1781 gst_event_parse_seek (event, &rate, &format, &flags,
1782 &cur_type, &cur, &stop_type, &stop);
1784 /* no negative rates yet */
1788 /* we need TIME format */
1789 if (format != src->segment.format)
1792 GST_DEBUG_OBJECT (src, "doing seek without event");
1794 cur_type = GST_SEEK_TYPE_SET;
1795 stop_type = GST_SEEK_TYPE_SET;
1798 /* get flush flag */
1799 flush = flags & GST_SEEK_FLAG_FLUSH;
1800 skip = flags & GST_SEEK_FLAG_SKIP;
1802 /* now we need to make sure the streaming thread is stopped. We do this by
1803 * either sending a FLUSH_START event downstream which will cause the
1804 * streaming thread to stop with a WRONG_STATE.
1805 * For a non-flushing seek we simply pause the task, which will happen as soon
1806 * as it completes one iteration (and thus might block when the sink is
1807 * blocking in preroll). */
1809 GST_DEBUG_OBJECT (src, "starting flush");
1810 gst_rtspsrc_flush (src, TRUE, FALSE);
1813 gst_task_pause (src->task);
1817 /* we should now be able to grab the streaming thread because we stopped it
1818 * with the above flush/pause code */
1819 GST_RTSP_STREAM_LOCK (src);
1821 GST_DEBUG_OBJECT (src, "stopped streaming");
1823 /* copy segment, we need this because we still need the old
1824 * segment when we close the current segment. */
1825 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
1827 /* configure the seek parameters in the seeksegment. We will then have the
1828 * right values in the segment to perform the seek */
1830 GST_DEBUG_OBJECT (src, "configuring seek");
1831 gst_segment_do_seek (&seeksegment, rate, format, flags,
1832 cur_type, cur, stop_type, stop, &update);
1835 /* figure out the last position we need to play. If it's configured (stop !=
1836 * -1), use that, else we play until the total duration of the file */
1837 if ((stop = seeksegment.stop) == -1)
1838 stop = seeksegment.duration;
1840 playing = (src->state == GST_RTSP_STATE_PLAYING);
1842 /* if we were playing, pause first */
1844 /* obtain current position in case seek fails */
1845 gst_rtspsrc_get_position (src);
1846 gst_rtspsrc_pause (src, FALSE, FALSE);
1849 gst_rtspsrc_do_seek (src, &seeksegment);
1851 /* and continue playing */
1853 gst_rtspsrc_play (src, &seeksegment, FALSE);
1855 /* prepare for streaming again */
1857 /* if we started flush, we stop now */
1858 GST_DEBUG_OBJECT (src, "stopping flush");
1859 gst_rtspsrc_flush (src, FALSE, playing);
1862 /* now we did the seek and can activate the new segment values */
1863 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
1865 /* if we're doing a segment seek, post a SEGMENT_START message */
1866 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
1867 gst_element_post_message (GST_ELEMENT_CAST (src),
1868 gst_message_new_segment_start (GST_OBJECT_CAST (src),
1869 src->segment.format, src->segment.position));
1872 /* now create the newsegment */
1873 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
1874 " to %" G_GINT64_FORMAT, src->segment.position, stop);
1876 /* store the newsegment event so it can be sent from the streaming thread. */
1877 if (src->start_segment)
1878 gst_event_unref (src->start_segment);
1879 src->start_segment = gst_event_new_segment (&src->segment);
1882 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
1883 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1884 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1885 stream->discont = TRUE;
1889 GST_RTSP_STREAM_UNLOCK (src);
1896 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
1901 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
1907 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
1911 gboolean res = TRUE;
1914 src = GST_RTSPSRC_CAST (parent);
1916 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
1917 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
1919 switch (GST_EVENT_TYPE (event)) {
1920 case GST_EVENT_SEEK:
1921 res = gst_rtspsrc_perform_seek (src, event);
1925 case GST_EVENT_NAVIGATION:
1926 case GST_EVENT_LATENCY:
1934 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
1935 res = gst_pad_send_event (target, event);
1936 gst_object_unref (target);
1938 gst_event_unref (event);
1941 gst_event_unref (event);
1947 /* this is the final event function we receive on the internal source pad when
1948 * we deal with TCP connections */
1950 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
1955 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
1957 switch (GST_EVENT_TYPE (event)) {
1958 case GST_EVENT_SEEK:
1960 case GST_EVENT_NAVIGATION:
1961 case GST_EVENT_LATENCY:
1963 gst_event_unref (event);
1970 /* this is the final query function we receive on the internal source pad when
1971 * we deal with TCP connections */
1973 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
1977 gboolean res = TRUE;
1979 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
1981 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
1982 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
1984 switch (GST_QUERY_TYPE (query)) {
1985 case GST_QUERY_POSITION:
1990 case GST_QUERY_DURATION:
1994 gst_query_parse_duration (query, &format, NULL);
1997 case GST_FORMAT_TIME:
1998 gst_query_set_duration (query, format, src->segment.duration);
2006 case GST_QUERY_LATENCY:
2008 /* we are live with a min latency of 0 and unlimited max latency, this
2009 * result will be updated by the session manager if there is any. */
2010 gst_query_set_latency (query, TRUE, 0, -1);
2020 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2022 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2026 gboolean res = FALSE;
2028 src = GST_RTSPSRC_CAST (parent);
2030 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2031 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2033 switch (GST_QUERY_TYPE (query)) {
2034 case GST_QUERY_DURATION:
2038 gst_query_parse_duration (query, &format, NULL);
2041 case GST_FORMAT_TIME:
2042 gst_query_set_duration (query, format, src->segment.duration);
2050 case GST_QUERY_SEEKING:
2054 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2055 if (format == GST_FORMAT_TIME) {
2057 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2059 /* seeking without duration is unlikely */
2060 seekable = seekable && src->seekable && src->segment.duration &&
2061 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2063 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2064 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2065 src->segment.start, src->segment.stop);
2072 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2074 /* forward the query to the proxy target pad */
2076 res = gst_pad_query (target, query);
2077 gst_object_unref (target);
2086 /* callback for RTCP messages to be sent to the server when operating in TCP
2088 static GstFlowReturn
2089 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2092 GstRTSPStream *stream;
2093 GstFlowReturn res = GST_FLOW_OK;
2098 GstRTSPMessage message = { 0 };
2099 GstRTSPConnection *conn;
2101 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2102 src = stream->parent;
2104 gst_buffer_map (buffer, &map, GST_MAP_READ);
2108 gst_rtsp_message_init_data (&message, stream->channel[1]);
2110 /* lend the body data to the message */
2111 gst_rtsp_message_take_body (&message, data, size);
2113 if (stream->conninfo.connection)
2114 conn = stream->conninfo.connection;
2116 conn = src->conninfo.connection;
2118 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2119 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2120 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2122 /* and steal it away again because we will free it when unreffing the
2124 gst_rtsp_message_steal_body (&message, &data, &size);
2125 gst_rtsp_message_unset (&message);
2127 gst_buffer_unmap (buffer, &map);
2128 gst_buffer_unref (buffer);
2133 static GstPadProbeReturn
2134 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2136 GstRTSPSrc *src = user_data;
2138 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2139 GST_DEBUG_PAD_NAME (pad));
2141 /* activate the streams */
2142 GST_OBJECT_LOCK (src);
2143 if (!src->need_activate)
2146 src->need_activate = FALSE;
2147 GST_OBJECT_UNLOCK (src);
2149 gst_rtspsrc_activate_streams (src);
2151 return GST_PAD_PROBE_OK;
2155 GST_OBJECT_UNLOCK (src);
2156 return GST_PAD_PROBE_OK;
2160 /* this callback is called when the session manager generated a new src pad with
2161 * payloaded RTP packets. We simply ghost the pad here. */
2163 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2166 GstPadTemplate *template;
2169 GstRTSPStream *stream;
2172 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2174 GST_RTSP_STATE_LOCK (src);
2176 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2177 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2178 goto unknown_stream;
2180 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
2182 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2184 goto unknown_stream;
2186 /* create a new pad we will use to stream to */
2187 template = gst_static_pad_template_get (&rtptemplate);
2188 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2189 gst_object_unref (template);
2192 stream->added = TRUE;
2193 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2194 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2195 gst_pad_set_active (stream->srcpad, TRUE);
2196 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2198 /* check if we added all streams */
2200 for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
2201 stream = (GstRTSPStream *) lstream->data;
2203 GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
2204 stream, stream->container, stream->disabled, stream->added);
2206 /* a container stream only needs one pad added. Also disabled streams don't
2208 if (!stream->container && !stream->disabled && !stream->added) {
2213 GST_RTSP_STATE_UNLOCK (src);
2216 GST_DEBUG_OBJECT (src, "We added all streams");
2217 /* when we get here, all stream are added and we can fire the no-more-pads
2219 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2227 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2228 GST_RTSP_STATE_UNLOCK (src);
2235 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2237 GstRTSPStream *stream;
2240 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2242 GST_RTSP_STATE_LOCK (src);
2243 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2245 goto unknown_stream;
2247 caps = stream->caps;
2249 gst_caps_ref (caps);
2250 GST_RTSP_STATE_UNLOCK (src);
2256 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2257 GST_RTSP_STATE_UNLOCK (src);
2263 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2265 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2271 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos (), TRUE);
2277 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2283 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2285 GstRTSPSrc *src = stream->parent;
2287 GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
2289 gst_rtspsrc_do_stream_eos (src, stream);
2293 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2295 GstRTSPSrc *src = stream->parent;
2297 GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
2299 gst_rtspsrc_do_stream_eos (src, stream);
2303 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2305 GstRTSPStream *stream;
2307 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2309 /* get stream for session */
2310 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2312 gst_rtspsrc_do_stream_eos (src, stream);
2317 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2319 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2323 /* try to get and configure a manager */
2325 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2326 GstRTSPTransport * transport)
2328 const gchar *manager;
2330 GstStateChangeReturn ret;
2332 /* find a manager */
2333 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2337 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2339 /* configure the manager */
2340 if (src->manager == NULL) {
2341 GObjectClass *klass;
2344 if (!(src->manager = gst_element_factory_make (manager, NULL))) {
2346 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
2350 goto use_no_manager;
2352 if (!(src->manager = gst_element_factory_make (manager, NULL)))
2353 goto manager_failed;
2356 /* we manage this element */
2357 gst_bin_add (GST_BIN_CAST (src), src->manager);
2359 GST_OBJECT_LOCK (src);
2360 target = GST_STATE_TARGET (src);
2361 GST_OBJECT_UNLOCK (src);
2363 ret = gst_element_set_state (src->manager, target);
2364 if (ret == GST_STATE_CHANGE_FAILURE)
2365 goto start_manager_failure;
2367 g_object_set (src->manager, "latency", src->latency, NULL);
2369 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2370 if (g_object_class_find_property (klass, "buffer-mode")) {
2371 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2372 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2374 gboolean need_slave;
2376 const gchar *encoding;
2378 /* buffer mode pauses are handled by adding offsets to buffer times,
2379 * but some depayloaders may have a hard time syncing output times
2380 * with such input times, e.g. container ones, most notably ASF */
2381 /* TODO alternatives are having an event that indicates these shifts,
2382 * or having rtsp extensions provide suggestion on buffer mode */
2383 need_slave = stream->container;
2384 if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
2385 (encoding = gst_structure_get_string (s, "encoding-name")))
2386 need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
2387 GST_DEBUG_OBJECT (src, "auto buffering mode, need_slave %d",
2389 /* valid duration implies not likely live pipeline,
2390 * so slaving in jitterbuffer does not make much sense
2391 * (and might mess things up due to bursts) */
2392 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
2393 src->segment.duration && !need_slave) {
2394 GST_DEBUG_OBJECT (src, "selected buffer");
2395 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER,
2398 GST_DEBUG_OBJECT (src, "selected slave");
2399 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2404 /* connect to signals if we did not already do so */
2405 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
2407 src->manager_sig_id =
2408 g_signal_connect (src->manager, "pad-added",
2409 (GCallback) new_manager_pad, src);
2410 src->manager_ptmap_id =
2411 g_signal_connect (src->manager, "request-pt-map",
2412 (GCallback) request_pt_map, src);
2414 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
2418 /* we stream directly to the manager, get some pads. Each RTSP stream goes
2419 * into a separate RTP session. */
2420 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
2421 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
2423 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
2424 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
2427 /* now configure the bandwidth in the manager */
2428 if (g_signal_lookup ("get-internal-session",
2429 G_OBJECT_TYPE (src->manager)) != 0) {
2430 GObject *rtpsession;
2432 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
2435 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
2437 stream->session = rtpsession;
2439 if (stream->as_bandwidth != -1) {
2440 GST_INFO_OBJECT (src, "setting AS: %f",
2441 (gdouble) (stream->as_bandwidth * 1000));
2442 g_object_set (rtpsession, "bandwidth",
2443 (gdouble) (stream->as_bandwidth * 1000), NULL);
2445 if (stream->rr_bandwidth != -1) {
2446 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
2447 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
2450 if (stream->rs_bandwidth != -1) {
2451 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
2452 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
2455 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2457 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
2459 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
2461 g_signal_connect (rtpsession, "on-ssrc-active",
2462 (GCallback) on_ssrc_active, stream);
2473 GST_DEBUG_OBJECT (src, "cannot get a session manager");
2478 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
2481 start_manager_failure:
2483 GST_DEBUG_OBJECT (src, "could not start session manager");
2488 /* free the UDP sources allocated when negotiating a transport.
2489 * This function is called when the server negotiated to a transport where the
2490 * UDP sources are not needed anymore, such as TCP or multicast. */
2492 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
2496 for (i = 0; i < 2; i++) {
2497 if (stream->udpsrc[i]) {
2498 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
2499 gst_object_unref (stream->udpsrc[i]);
2500 stream->udpsrc[i] = NULL;
2505 /* for TCP, create pads to send and receive data to and from the manager and to
2506 * intercept various events and queries
2509 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
2510 GstRTSPTransport * transport, GstPad ** outpad)
2513 GstPadTemplate *template;
2514 GstPad *pad0, *pad1;
2516 /* configure for interleaved delivery, nothing needs to be done
2517 * here, the loop function will call the chain functions of the
2518 * session manager. */
2519 stream->channel[0] = transport->interleaved.min;
2520 stream->channel[1] = transport->interleaved.max;
2521 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
2522 stream->channel[0], stream->channel[1]);
2524 /* we can remove the allocated UDP ports now */
2525 gst_rtspsrc_stream_free_udp (stream);
2527 /* no session manager, send data to srcpad directly */
2528 if (!stream->channelpad[0]) {
2529 GST_DEBUG_OBJECT (src, "no manager, creating pad");
2531 /* create a new pad we will use to stream to */
2532 name = g_strdup_printf ("stream_%u", stream->id);
2533 template = gst_static_pad_template_get (&rtptemplate);
2534 stream->channelpad[0] = gst_pad_new_from_template (template, name);
2535 gst_object_unref (template);
2538 /* set caps and activate */
2539 gst_pad_use_fixed_caps (stream->channelpad[0]);
2540 gst_pad_set_active (stream->channelpad[0], TRUE);
2542 *outpad = gst_object_ref (stream->channelpad[0]);
2544 GST_DEBUG_OBJECT (src, "using manager source pad");
2546 template = gst_static_pad_template_get (&anysrctemplate);
2548 /* allocate pads for sending the channel data into the manager */
2549 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
2550 gst_pad_link (pad0, stream->channelpad[0]);
2551 gst_object_unref (stream->channelpad[0]);
2552 stream->channelpad[0] = pad0;
2553 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
2554 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
2555 gst_pad_set_element_private (pad0, src);
2556 gst_pad_set_active (pad0, TRUE);
2558 if (stream->channelpad[1]) {
2559 /* if we have a sinkpad for the other channel, create a pad and link to the
2561 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
2562 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
2563 gst_pad_link (pad1, stream->channelpad[1]);
2564 gst_object_unref (stream->channelpad[1]);
2565 stream->channelpad[1] = pad1;
2566 gst_pad_set_active (pad1, TRUE);
2568 gst_object_unref (template);
2570 /* setup RTCP transport back to the server if we have to. */
2571 if (src->manager && src->do_rtcp) {
2574 template = gst_static_pad_template_get (&anysinktemplate);
2576 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
2577 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
2578 gst_pad_set_element_private (stream->rtcppad, stream);
2579 gst_pad_set_active (stream->rtcppad, TRUE);
2581 /* get session RTCP pad */
2582 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2583 pad = gst_element_get_request_pad (src->manager, name);
2588 gst_pad_link (pad, stream->rtcppad);
2589 gst_object_unref (pad);
2592 gst_object_unref (template);
2598 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
2599 GstRTSPTransport * transport, const gchar ** destination, gint * min,
2600 gint * max, guint * ttl)
2602 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2604 if (!(*destination = transport->destination))
2605 *destination = stream->destination;
2608 /* transport first */
2609 *min = transport->port.min;
2610 *max = transport->port.max;
2611 if (*min == -1 && *max == -1) {
2612 /* then try from SDP */
2613 if (stream->port != 0) {
2614 *min = stream->port;
2615 *max = stream->port + 1;
2621 if (!(*ttl = transport->ttl))
2626 /* first take the source, then the endpoint to figure out where to send
2628 if (!(*destination = transport->source)) {
2629 if (src->conninfo.connection)
2630 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
2631 else if (stream->conninfo.connection)
2633 gst_rtsp_connection_get_ip (stream->conninfo.connection);
2637 /* for unicast we only expect the ports here */
2638 *min = transport->server_port.min;
2639 *max = transport->server_port.max;
2644 /* For multicast create UDP sources and join the multicast group. */
2646 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
2647 GstRTSPTransport * transport, GstPad ** outpad)
2650 const gchar *destination;
2653 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
2655 /* we can remove the allocated UDP ports now */
2656 gst_rtspsrc_stream_free_udp (stream);
2658 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
2661 /* we need a destination now */
2662 if (destination == NULL)
2663 goto no_destination;
2665 /* we really need ports now or we won't be able to receive anything at all */
2666 if (min == -1 && max == -1)
2669 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
2670 destination, min, max);
2672 /* creating UDP source for RTP */
2674 uri = g_strdup_printf ("udp://%s:%d", destination, min);
2675 stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2677 if (stream->udpsrc[0] == NULL)
2680 /* take ownership */
2681 gst_object_ref_sink (stream->udpsrc[0]);
2684 gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
2687 /* creating another UDP source for RTCP */
2689 uri = g_strdup_printf ("udp://%s:%d", destination, max);
2690 stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2692 if (stream->udpsrc[1] == NULL)
2695 /* take ownership */
2696 gst_object_ref_sink (stream->udpsrc[1]);
2698 gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
2705 GST_DEBUG_OBJECT (src, "no UDP source element found");
2710 GST_DEBUG_OBJECT (src, "no destination found");
2715 GST_DEBUG_OBJECT (src, "no ports found");
2720 /* configure the remainder of the UDP ports */
2722 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
2723 GstRTSPTransport * transport, GstPad ** outpad)
2725 /* we manage the UDP elements now. For unicast, the UDP sources where
2726 * allocated in the stream when we suggested a transport. */
2727 if (stream->udpsrc[0]) {
2728 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
2730 GST_DEBUG_OBJECT (src, "setting up UDP source");
2732 /* configure a timeout on the UDP port. When the timeout message is
2733 * posted, we assume UDP transport is not possible. We reconnect using TCP
2735 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
2738 /* get output pad of the UDP source. */
2739 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
2741 /* save it so we can unblock */
2742 stream->blockedpad = *outpad;
2744 /* configure pad block on the pad. As soon as there is dataflow on the
2745 * UDP source, we know that UDP is not blocked by a firewall and we can
2746 * configure all the streams to let the application autoplug decoders. */
2748 gst_pad_add_probe (stream->blockedpad,
2749 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
2751 if (stream->channelpad[0]) {
2752 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
2753 /* configure for UDP delivery, we need to connect the UDP pads to
2754 * the session plugin. */
2755 gst_pad_link (*outpad, stream->channelpad[0]);
2756 gst_object_unref (*outpad);
2758 /* we connected to pad-added signal to get pads from the manager */
2760 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
2765 if (stream->udpsrc[1]) {
2766 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
2768 if (stream->channelpad[1]) {
2771 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
2773 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
2774 gst_pad_link (pad, stream->channelpad[1]);
2775 gst_object_unref (pad);
2777 /* leave unlinked */
2783 /* configure the UDP sink back to the server for status reports */
2785 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
2786 GstRTSPStream * stream, GstRTSPTransport * transport)
2789 gint rtp_port, rtcp_port;
2790 gboolean do_rtp, do_rtcp;
2791 const gchar *destination;
2796 /* get transport info */
2797 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
2798 &rtp_port, &rtcp_port, &ttl);
2800 /* see what we need to do */
2801 do_rtp = (rtp_port != -1);
2802 /* it's possible that the server does not want us to send RTCP in which case
2804 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
2806 /* we need a destination when we have RTP or RTCP ports */
2807 if (destination == NULL && (do_rtp || do_rtcp))
2808 goto no_destination;
2810 /* try to construct the fakesrc to the RTP port of the server to open up any
2813 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
2816 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
2817 stream->udpsink[0] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2819 if (stream->udpsink[0] == NULL)
2820 goto no_sink_element;
2822 /* don't join multicast group, we will have the source socket do that */
2823 /* no sync or async state changes needed */
2824 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
2825 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2827 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2829 if (stream->udpsrc[0]) {
2830 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
2831 * so that NAT firewalls will open a hole for us */
2832 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
2833 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
2834 /* configure socket and make sure udpsink does not close it when shutting
2835 * down, it belongs to udpsrc after all. */
2836 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
2837 "close-socket", FALSE, NULL);
2838 g_object_unref (socket);
2841 /* the source for the dummy packets to open up NAT */
2842 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
2843 if (stream->fakesrc == NULL)
2844 goto no_fakesrc_element;
2846 /* random data in 5 buffers, a size of 200 bytes should be fine */
2847 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
2848 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
2850 /* we don't want to consider this a sink */
2851 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
2853 /* keep everything locked */
2854 gst_element_set_locked_state (stream->udpsink[0], TRUE);
2855 gst_element_set_locked_state (stream->fakesrc, TRUE);
2857 gst_object_ref (stream->udpsink[0]);
2858 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
2859 gst_object_ref (stream->fakesrc);
2860 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
2862 gst_element_link (stream->fakesrc, stream->udpsink[0]);
2865 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
2868 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
2869 stream->udpsink[1] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2871 if (stream->udpsink[1] == NULL)
2872 goto no_sink_element;
2874 /* don't join multicast group, we will have the source socket do that */
2875 /* no sync or async state changes needed */
2876 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
2877 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2879 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2881 if (stream->udpsrc[1]) {
2882 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
2883 * because some servers check the port number of where it sends RTCP to identify
2884 * the RTCP packets it receives */
2885 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
2886 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
2887 /* configure socket and make sure udpsink does not close it when shutting
2888 * down, it belongs to udpsrc after all. */
2889 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
2890 "close-socket", FALSE, NULL);
2891 g_object_unref (socket);
2894 /* we don't want to consider this a sink */
2895 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
2897 /* we keep this playing always */
2898 gst_element_set_locked_state (stream->udpsink[1], TRUE);
2899 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
2901 gst_object_ref (stream->udpsink[1]);
2902 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
2904 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
2906 /* get session RTCP pad */
2907 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2908 pad = gst_element_get_request_pad (src->manager, name);
2913 gst_pad_link (pad, stream->rtcppad);
2914 gst_object_unref (pad);
2923 GST_DEBUG_OBJECT (src, "no destination address specified");
2928 GST_DEBUG_OBJECT (src, "no UDP sink element found");
2933 GST_DEBUG_OBJECT (src, "no fakesrc element found");
2938 /* sets up all elements needed for streaming over the specified transport.
2939 * Does not yet expose the element pads, this will be done when there is actuall
2940 * dataflow detected, which might never happen when UDP is blocked in a
2941 * firewall, for example.
2944 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
2945 GstRTSPTransport * transport)
2948 GstPad *outpad = NULL;
2949 GstPadTemplate *template;
2954 src = stream->parent;
2956 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
2958 s = gst_caps_get_structure (stream->caps, 0);
2960 /* get the proper mime type for this stream now */
2961 if (gst_rtsp_transport_get_mime (transport->trans, &mime) < 0)
2962 goto unknown_transport;
2964 goto unknown_transport;
2966 /* configure the final mime type */
2967 GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
2968 gst_structure_set_name (s, mime);
2970 /* try to get and configure a manager, channelpad[0-1] will be configured with
2971 * the pads for the manager, or NULL when no manager is needed. */
2972 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
2975 switch (transport->lower_transport) {
2976 case GST_RTSP_LOWER_TRANS_TCP:
2977 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
2978 goto transport_failed;
2980 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2981 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
2982 goto transport_failed;
2983 /* fallthrough, the rest is the same for UDP and MCAST */
2984 case GST_RTSP_LOWER_TRANS_UDP:
2985 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
2986 goto transport_failed;
2987 /* configure udpsinks back to the server for RTCP messages and for the
2988 * dummy RTP messages to open NAT. */
2989 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
2990 goto transport_failed;
2993 goto unknown_transport;
2997 GST_DEBUG_OBJECT (src, "creating ghostpad");
2999 gst_pad_use_fixed_caps (outpad);
3001 /* create ghostpad, don't add just yet, this will be done when we activate
3003 name = g_strdup_printf ("stream_%u", stream->id);
3004 template = gst_static_pad_template_get (&rtptemplate);
3005 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3006 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3007 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3008 gst_object_unref (template);
3011 gst_object_unref (outpad);
3013 /* mark pad as ok */
3014 stream->last_ret = GST_FLOW_OK;
3021 GST_DEBUG_OBJECT (src, "failed to configure transport");
3026 GST_DEBUG_OBJECT (src, "unknown transport");
3031 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3036 /* send a couple of dummy random packets on the receiver RTP port to the server,
3037 * this should make a firewall think we initiated the data transfer and
3038 * hopefully allow packets to go from the sender port to our RTP receiver port */
3040 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3044 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3047 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3048 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3050 if (stream->fakesrc && stream->udpsink[0]) {
3051 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3052 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3053 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3054 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3055 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3061 /* Adds the source pads of all configured streams to the element.
3062 * This code is performed when we detected dataflow.
3064 * We detect dataflow from either the _loop function or with pad probes on the
3068 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3072 GST_DEBUG_OBJECT (src, "activating streams");
3074 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3075 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3077 if (stream->udpsrc[0]) {
3078 /* remove timeout, we are streaming now and timeouts will be handled by
3079 * the session manager and jitter buffer */
3080 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3082 if (stream->srcpad) {
3083 /* if we don't have a session manager, set the caps now. If we have a
3084 * session, we will get a notification of the pad and the caps. */
3085 if (!src->manager) {
3086 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3087 gst_pad_set_caps (stream->srcpad, stream->caps);
3090 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3091 gst_pad_set_active (stream->srcpad, TRUE);
3093 if (!stream->added) {
3094 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3095 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3096 stream->added = TRUE;
3101 /* unblock all pads */
3102 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3103 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3105 if (stream->blockid) {
3106 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3107 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3108 stream->blockid = 0;
3116 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
3119 guint64 start, stop;
3120 gdouble play_speed, play_scale;
3122 GST_DEBUG_OBJECT (src, "configuring stream caps");
3124 start = segment->position;
3125 stop = segment->duration;
3126 play_speed = segment->rate;
3127 play_scale = segment->applied_rate;
3129 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3130 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3133 if ((caps = stream->caps)) {
3134 caps = gst_caps_make_writable (caps);
3136 if (stream->timebase != -1)
3137 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3138 (guint) stream->timebase, NULL);
3139 if (stream->seqbase != -1)
3140 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3141 (guint) stream->seqbase, NULL);
3142 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3144 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3145 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3146 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3148 stream->caps = caps;
3150 GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
3153 GST_DEBUG_OBJECT (src, "clear session");
3154 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3158 static GstFlowReturn
3159 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3164 /* store the value */
3165 stream->last_ret = ret;
3167 /* if it's success we can return the value right away */
3168 if (ret == GST_FLOW_OK)
3171 /* any other error that is not-linked can be returned right
3173 if (ret != GST_FLOW_NOT_LINKED)
3176 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3177 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3178 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3180 ret = ostream->last_ret;
3181 /* some other return value (must be SUCCESS but we can return
3182 * other values as well) */
3183 if (ret != GST_FLOW_NOT_LINKED)
3186 /* if we get here, all other pads were unlinked and we return
3187 * NOT_LINKED then */
3193 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3194 GstEvent * event, gboolean source)
3196 gboolean res = TRUE;
3198 /* only streams that have a connection to the outside world */
3199 if (stream->srcpad == NULL)
3202 if (source && stream->udpsrc[0]) {
3203 gst_event_ref (event);
3204 res = gst_element_send_event (stream->udpsrc[0], event);
3205 } else if (stream->channelpad[0]) {
3206 gst_event_ref (event);
3207 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3208 res = gst_pad_push_event (stream->channelpad[0], event);
3210 res = gst_pad_send_event (stream->channelpad[0], event);
3213 if (source && stream->udpsrc[1]) {
3214 gst_event_ref (event);
3215 res &= gst_element_send_event (stream->udpsrc[1], event);
3216 } else if (stream->channelpad[1]) {
3217 gst_event_ref (event);
3218 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3219 res &= gst_pad_push_event (stream->channelpad[1], event);
3221 res &= gst_pad_send_event (stream->channelpad[1], event);
3225 gst_event_unref (event);
3231 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
3234 gboolean res = TRUE;
3236 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3237 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3239 gst_event_ref (event);
3240 res &= gst_rtspsrc_stream_push_event (src, ostream, event, source);
3242 gst_event_unref (event);
3247 static GstRTSPResult
3248 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3253 if (info->connection == NULL) {
3254 if (info->url == NULL) {
3255 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
3256 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
3260 /* create connection */
3261 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
3262 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
3263 goto could_not_create;
3266 g_free (info->url_str);
3267 info->url_str = gst_rtsp_url_get_request_uri (info->url);
3269 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
3271 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
3272 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
3274 if (src->proxy_host) {
3275 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
3277 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
3282 if (!info->connected) {
3285 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
3286 ("Connecting to %s", info->location));
3287 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
3289 gst_rtsp_connection_connect (info->connection,
3290 src->ptcp_timeout)) < 0)
3291 goto could_not_connect;
3293 info->connected = TRUE;
3300 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
3305 gchar *str = gst_rtsp_strresult (res);
3306 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
3312 gchar *str = gst_rtsp_strresult (res);
3313 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
3319 static GstRTSPResult
3320 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
3323 if (info->connected) {
3324 GST_DEBUG_OBJECT (src, "closing connection...");
3325 gst_rtsp_connection_close (info->connection);
3326 info->connected = FALSE;
3328 if (free && info->connection) {
3329 /* free connection */
3330 GST_DEBUG_OBJECT (src, "freeing connection...");
3331 gst_rtsp_connection_free (info->connection);
3332 info->connection = NULL;
3337 static GstRTSPResult
3338 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3343 GST_DEBUG_OBJECT (src, "reconnecting connection...");
3344 gst_rtsp_conninfo_close (src, info, FALSE);
3345 res = gst_rtsp_conninfo_connect (src, info, async);
3351 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
3355 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
3356 if (src->conninfo.connection) {
3357 GST_DEBUG_OBJECT (src, "connection flush");
3358 gst_rtsp_connection_flush (src->conninfo.connection, flush);
3360 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3361 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3362 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
3363 if (stream->conninfo.connection)
3364 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
3368 /* FIXME, handle server request, reply with OK, for now */
3369 static GstRTSPResult
3370 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
3371 GstRTSPMessage * request)
3373 GstRTSPMessage response = { 0 };
3376 GST_DEBUG_OBJECT (src, "got server request message");
3379 gst_rtsp_message_dump (request);
3381 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
3383 if (res == GST_RTSP_ENOTIMPL) {
3384 /* default implementation, send OK */
3386 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
3391 GST_DEBUG_OBJECT (src, "replying with OK");
3394 gst_rtsp_message_dump (&response);
3396 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
3400 gst_rtsp_message_unset (&response);
3401 } else if (res == GST_RTSP_EEOF)
3409 gst_rtsp_message_unset (&response);
3414 /* send server keep-alive */
3415 static GstRTSPResult
3416 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
3418 GstRTSPMessage request = { 0 };
3420 GstRTSPMethod method;
3423 GST_DEBUG_OBJECT (src, "creating server keep-alive");
3425 /* find a method to use for keep-alive */
3426 if (src->methods & GST_RTSP_GET_PARAMETER)
3427 method = GST_RTSP_GET_PARAMETER;
3429 method = GST_RTSP_OPTIONS;
3432 control = src->control;
3434 control = src->conninfo.url_str;
3436 if (control == NULL)
3439 res = gst_rtsp_message_init_request (&request, method, control);
3444 gst_rtsp_message_dump (&request);
3447 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
3452 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
3453 gst_rtsp_message_unset (&request);
3460 GST_WARNING_OBJECT (src, "no control url to send keepalive");
3465 gchar *str = gst_rtsp_strresult (res);
3467 gst_rtsp_message_unset (&request);
3468 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
3469 ("Could not send keep-alive. (%s)", str));
3475 static GstFlowReturn
3476 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
3478 GstRTSPMessage message = { 0 };
3481 GstRTSPStream *stream;
3482 GstPad *outpad = NULL;
3485 GstFlowReturn ret = GST_FLOW_OK;
3487 gboolean is_rtcp, have_data;
3489 /* here we are only interested in data messages */
3492 GTimeVal tv_timeout;
3494 /* get the next timeout interval */
3495 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3497 /* see if the timeout period expired */
3498 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
3499 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
3500 /* send keep-alive, only act on interrupt, a warning will be posted for
3502 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3504 /* get new timeout */
3505 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3508 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
3509 tv_timeout.tv_sec, tv_timeout.tv_usec);
3511 /* protect the connection with the connection lock so that we can see when
3512 * we are finished doing server communication */
3514 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3515 &message, src->ptcp_timeout);
3519 GST_DEBUG_OBJECT (src, "we received a server message");
3521 case GST_RTSP_EINTR:
3522 /* we got interrupted this means we need to stop */
3524 case GST_RTSP_ETIMEOUT:
3525 /* no reply, send keep alive */
3526 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3527 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3531 /* go EOS when the server closed the connection */
3537 switch (message.type) {
3538 case GST_RTSP_MESSAGE_REQUEST:
3539 /* server sends us a request message, handle it */
3541 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3543 if (res == GST_RTSP_EEOF)
3546 goto handle_request_failed;
3548 case GST_RTSP_MESSAGE_RESPONSE:
3549 /* we ignore response messages */
3550 GST_DEBUG_OBJECT (src, "ignoring response message");
3552 gst_rtsp_message_dump (&message);
3554 case GST_RTSP_MESSAGE_DATA:
3555 GST_DEBUG_OBJECT (src, "got data message");
3559 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3566 channel = message.type_data.data.channel;
3568 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
3570 goto unknown_stream;
3572 if (channel == stream->channel[0]) {
3573 outpad = stream->channelpad[0];
3575 } else if (channel == stream->channel[1]) {
3576 outpad = stream->channelpad[1];
3582 /* take a look at the body to figure out what we have */
3583 gst_rtsp_message_get_body (&message, &data, &size);
3585 goto invalid_length;
3587 /* channels are not correct on some servers, do extra check */
3588 if (data[1] >= 200 && data[1] <= 204) {
3589 /* hmm RTCP message switch to the RTCP pad of the same stream. */
3590 outpad = stream->channelpad[1];
3594 /* we have no clue what this is, just ignore then. */
3596 goto unknown_stream;
3598 /* take the message body for further processing */
3599 gst_rtsp_message_steal_body (&message, &data, &size);
3601 /* strip the trailing \0 */
3604 buf = gst_buffer_new ();
3605 gst_buffer_take_memory (buf, -1,
3606 gst_memory_new_wrapped (0, data, g_free, size, 0, size));
3608 /* don't need message anymore */
3609 gst_rtsp_message_unset (&message);
3611 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
3614 if (src->need_activate) {
3615 gst_rtspsrc_activate_streams (src);
3616 src->need_activate = FALSE;
3619 if (src->base_time == -1) {
3620 /* Take current running_time. This timestamp will be put on
3621 * the first buffer of each stream because we are a live source and so we
3622 * timestamp with the running_time. When we are dealing with TCP, we also
3623 * only timestamp the first buffer (using the DISCONT flag) because a server
3624 * typically bursts data, for which we don't want to compensate by speeding
3625 * up the media. The other timestamps will be interpollated from this one
3626 * using the RTP timestamps. */
3627 GST_OBJECT_LOCK (src);
3628 if (GST_ELEMENT_CLOCK (src)) {
3630 GstClockTime base_time;
3632 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
3633 base_time = GST_ELEMENT_CAST (src)->base_time;
3635 src->base_time = now - base_time;
3637 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
3638 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
3640 GST_OBJECT_UNLOCK (src);
3643 if (stream->discont && !is_rtcp) {
3644 /* mark first RTP buffer as discont */
3645 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
3646 stream->discont = FALSE;
3647 /* first buffer gets the timestamp, other buffers are not timestamped and
3648 * their presentation time will be interpollated from the rtp timestamps. */
3649 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
3650 GST_TIME_ARGS (src->base_time));
3652 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
3655 /* chain to the peer pad */
3656 if (GST_PAD_IS_SINK (outpad))
3657 ret = gst_pad_chain (outpad, buf);
3659 ret = gst_pad_push (outpad, buf);
3662 /* combine all stream flows for the data transport */
3663 ret = gst_rtspsrc_combine_flows (src, stream, ret);
3670 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
3671 gst_rtsp_message_unset (&message);
3676 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3677 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3678 ("The server closed the connection."));
3679 src->conninfo.connected = FALSE;
3680 gst_rtsp_message_unset (&message);
3681 return GST_FLOW_EOS;
3685 gst_rtsp_message_unset (&message);
3686 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3687 gst_rtspsrc_connection_flush (src, FALSE);
3688 return GST_FLOW_FLUSHING;
3692 gchar *str = gst_rtsp_strresult (res);
3694 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3695 ("Could not receive message. (%s)", str));
3698 gst_rtsp_message_unset (&message);
3699 return GST_FLOW_ERROR;
3701 handle_request_failed:
3703 gchar *str = gst_rtsp_strresult (res);
3705 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3706 ("Could not handle server message. (%s)", str));
3708 gst_rtsp_message_unset (&message);
3709 return GST_FLOW_ERROR;
3713 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3714 ("Short message received, ignoring."));
3715 gst_rtsp_message_unset (&message);
3720 static GstFlowReturn
3721 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
3724 GstRTSPMessage message = { 0 };
3728 GTimeVal tv_timeout;
3730 /* get the next timeout interval */
3731 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3733 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
3734 (gint) tv_timeout.tv_sec);
3736 gst_rtsp_message_unset (&message);
3738 /* we should continue reading the TCP socket because the server might
3739 * send us requests. When the session timeout expires, we need to send a
3740 * keep-alive request to keep the session open. */
3741 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3742 &message, &tv_timeout);
3746 GST_DEBUG_OBJECT (src, "we received a server message");
3748 case GST_RTSP_EINTR:
3749 /* we got interrupted, see what we have to do */
3751 case GST_RTSP_ETIMEOUT:
3752 /* send keep-alive, ignore the result, a warning will be posted. */
3753 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3754 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3758 /* server closed the connection. not very fatal for UDP, reconnect and
3759 * see what happens. */
3760 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3761 ("The server closed the connection."));
3763 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
3771 switch (message.type) {
3772 case GST_RTSP_MESSAGE_REQUEST:
3773 /* server sends us a request message, handle it */
3775 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3777 if (res == GST_RTSP_EEOF)
3780 goto handle_request_failed;
3782 case GST_RTSP_MESSAGE_RESPONSE:
3783 /* we ignore response and data messages */
3784 GST_DEBUG_OBJECT (src, "ignoring response message");
3786 gst_rtsp_message_dump (&message);
3787 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
3788 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
3789 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
3790 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
3791 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3798 case GST_RTSP_MESSAGE_DATA:
3799 /* we ignore response and data messages */
3800 GST_DEBUG_OBJECT (src, "ignoring data message");
3803 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3809 /* we get here when the connection got interrupted */
3812 gst_rtsp_message_unset (&message);
3813 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3814 gst_rtspsrc_connection_flush (src, FALSE);
3815 return GST_FLOW_FLUSHING;
3819 gchar *str = gst_rtsp_strresult (res);
3822 src->conninfo.connected = FALSE;
3823 if (res != GST_RTSP_EINTR) {
3824 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
3825 ("Could not connect to server. (%s)", str));
3827 ret = GST_FLOW_ERROR;
3829 ret = GST_FLOW_FLUSHING;
3835 gchar *str = gst_rtsp_strresult (res);
3837 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3838 ("Could not receive message. (%s)", str));
3840 return GST_FLOW_ERROR;
3842 handle_request_failed:
3844 gchar *str = gst_rtsp_strresult (res);
3847 gst_rtsp_message_unset (&message);
3848 if (res != GST_RTSP_EINTR) {
3849 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3850 ("Could not handle server message. (%s)", str));
3852 ret = GST_FLOW_ERROR;
3854 ret = GST_FLOW_FLUSHING;
3860 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3861 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3862 ("The server closed the connection."));
3863 src->conninfo.connected = FALSE;
3864 gst_rtsp_message_unset (&message);
3865 return GST_FLOW_EOS;
3869 static GstRTSPResult
3870 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
3872 GstRTSPResult res = GST_RTSP_OK;
3875 GST_DEBUG_OBJECT (src, "doing reconnect");
3877 GST_OBJECT_LOCK (src);
3878 /* only restart when the pads were not yet activated, else we were
3879 * streaming over UDP */
3880 restart = src->need_activate;
3881 GST_OBJECT_UNLOCK (src);
3883 /* no need to restart, we're done */
3887 /* we can try only TCP now */
3888 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
3890 /* close and cleanup our state */
3891 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
3894 /* see if we have TCP left to try. Also don't try TCP when we were configured
3896 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
3899 /* We post a warning message now to inform the user
3900 * that nothing happened. It's most likely a firewall thing. */
3901 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3902 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3903 "firewall is blocking it. Retrying using a TCP connection.",
3904 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3906 /* open new connection using tcp */
3907 if (gst_rtspsrc_open (src, async) < 0)
3910 /* start playback */
3911 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
3920 src->cur_protocols = 0;
3921 /* no transport possible, post an error and stop */
3922 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3923 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3924 "firewall is blocking it. No other protocols to try.",
3925 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3926 return GST_FLOW_ERROR;
3930 GST_DEBUG_OBJECT (src, "open failed");
3935 GST_DEBUG_OBJECT (src, "play failed");
3941 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
3945 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
3948 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
3951 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
3954 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
3962 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
3966 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
3969 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
3972 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
3975 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
3983 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
3987 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
3990 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
3993 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
3996 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4004 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4008 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4011 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4014 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4017 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4025 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4027 if (ret == GST_RTSP_OK)
4028 gst_rtspsrc_loop_complete_cmd (src, cmd);
4029 else if (ret == GST_RTSP_EINTR)
4030 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4032 gst_rtspsrc_loop_error_cmd (src, cmd);
4036 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
4040 /* start new request */
4041 gst_rtspsrc_loop_start_cmd (src, cmd);
4043 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4045 GST_OBJECT_LOCK (src);
4046 old = src->loop_cmd;
4047 if (old != CMD_WAIT) {
4048 src->loop_cmd = CMD_WAIT;
4049 GST_OBJECT_UNLOCK (src);
4050 /* cancel previous request */
4051 gst_rtspsrc_loop_cancel_cmd (src, old);
4052 GST_OBJECT_LOCK (src);
4054 src->loop_cmd = cmd;
4055 /* interrupt if allowed */
4057 GST_DEBUG_OBJECT (src, "start connection flush");
4058 gst_rtspsrc_connection_flush (src, TRUE);
4061 gst_task_start (src->task);
4062 GST_OBJECT_UNLOCK (src);
4066 gst_rtspsrc_loop (GstRTSPSrc * src)
4070 if (!src->conninfo.connection || !src->conninfo.connected)
4073 if (src->interleaved)
4074 ret = gst_rtspsrc_loop_interleaved (src);
4076 ret = gst_rtspsrc_loop_udp (src);
4078 if (ret != GST_FLOW_OK)
4086 GST_WARNING_OBJECT (src, "we are not connected");
4087 ret = GST_FLOW_FLUSHING;
4092 const gchar *reason = gst_flow_get_name (ret);
4094 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4095 src->running = FALSE;
4096 if (ret == GST_FLOW_EOS) {
4097 /* perform EOS logic */
4098 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4099 gst_element_post_message (GST_ELEMENT_CAST (src),
4100 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4101 src->segment.format, src->segment.position));
4103 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4105 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4106 /* for fatal errors we post an error message, post the error before the
4107 * EOS so the app knows about the error first. */
4108 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4109 ("Internal data flow error."),
4110 ("streaming task paused, reason %s (%d)", reason, ret));
4111 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4117 #ifndef GST_DISABLE_GST_DEBUG
4118 static const gchar *
4119 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4123 while (method != 0) {
4140 static const gchar *
4141 gst_rtspsrc_skip_lws (const gchar * s)
4143 while (g_ascii_isspace (*s))
4148 static const gchar *
4149 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4151 while (s > start && g_ascii_isspace (*(s - 1)))
4156 static const gchar *
4157 gst_rtspsrc_skip_commas (const gchar * s)
4159 /* The grammar allows for multiple commas */
4160 while (g_ascii_isspace (*s) || *s == ',')
4165 static const gchar *
4166 gst_rtspsrc_skip_item (const gchar * s)
4168 gboolean quoted = FALSE;
4169 const gchar *start = s;
4171 /* A list item ends at the last non-whitespace character
4172 * before a comma which is not inside a quoted-string. Or at
4173 * the end of the string.
4179 if (*s == '\\' && *(s + 1))
4188 return gst_rtspsrc_unskip_lws (s, start);
4192 gst_rtsp_decode_quoted_string (gchar * quoted_string)
4196 src = quoted_string + 1;
4197 dst = quoted_string;
4198 while (*src && *src != '"') {
4199 if (*src == '\\' && *(src + 1))
4206 /* Extract the authentication tokens that the server provided for each method
4207 * into an array of structures and give those to the connection object.
4210 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
4211 const gchar * header, gboolean * stale)
4213 GSList *list = NULL, *iter;
4215 gchar *item, *eq, *name_end, *value;
4217 g_return_if_fail (stale != NULL);
4219 gst_rtsp_connection_clear_auth_params (conn);
4222 /* Parse a header whose content is described by RFC2616 as
4223 * "#something", where "something" does not itself contain commas,
4224 * except as part of quoted-strings, into a list of allocated strings.
4226 header = gst_rtspsrc_skip_commas (header);
4228 end = gst_rtspsrc_skip_item (header);
4229 list = g_slist_prepend (list, g_strndup (header, end - header));
4230 header = gst_rtspsrc_skip_commas (end);
4235 list = g_slist_reverse (list);
4236 for (iter = list; iter; iter = iter->next) {
4239 eq = strchr (item, '=');
4241 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
4242 if (name_end == item) {
4243 /* That's no good... */
4250 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
4252 gst_rtsp_decode_quoted_string (value);
4256 if (item && (strcmp (item, "stale") == 0) &&
4257 value && (strcmp (value, "TRUE") == 0))
4259 gst_rtsp_connection_set_auth_param (conn, item, value);
4263 g_slist_free (list);
4266 /* Parse a WWW-Authenticate Response header and determine the
4267 * available authentication methods
4269 * This code should also cope with the fact that each WWW-Authenticate
4270 * header can contain multiple challenge methods + tokens
4272 * At the moment, for Basic auth, we just do a minimal check and don't
4273 * even parse out the realm */
4275 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
4276 GstRTSPConnection * conn, gboolean * stale)
4280 g_return_if_fail (hdr != NULL);
4281 g_return_if_fail (methods != NULL);
4282 g_return_if_fail (stale != NULL);
4284 /* Skip whitespace at the start of the string */
4285 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
4287 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
4288 *methods |= GST_RTSP_AUTH_BASIC;
4289 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
4290 *methods |= GST_RTSP_AUTH_DIGEST;
4291 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
4296 * gst_rtspsrc_setup_auth:
4297 * @src: the rtsp source
4299 * Configure a username and password and auth method on the
4300 * connection object based on a response we received from the
4303 * Currently, this requires that a username and password were supplied
4304 * in the uri. In the future, they may be requested on demand by sending
4305 * a message up the bus.
4307 * Returns: TRUE if authentication information could be set up correctly.
4310 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
4314 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
4315 GstRTSPAuthMethod method;
4316 GstRTSPResult auth_result;
4318 GstRTSPConnection *conn;
4320 gboolean stale = FALSE;
4322 conn = src->conninfo.connection;
4324 /* Identify the available auth methods and see if any are supported */
4325 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
4326 &hdr, 0) == GST_RTSP_OK) {
4327 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
4330 if (avail_methods == GST_RTSP_AUTH_NONE)
4331 goto no_auth_available;
4333 /* For digest auth, if the response indicates that the session
4334 * data are stale, we just update them in the connection object and
4335 * return TRUE to retry the request */
4337 src->tried_url_auth = FALSE;
4339 url = gst_rtsp_connection_get_url (conn);
4341 /* Do we have username and password available? */
4342 if (url != NULL && !src->tried_url_auth && url->user != NULL
4343 && url->passwd != NULL) {
4346 src->tried_url_auth = TRUE;
4347 GST_DEBUG_OBJECT (src,
4348 "Attempting authentication using credentials from the URL");
4350 user = src->user_id;
4351 pass = src->user_pw;
4352 GST_DEBUG_OBJECT (src,
4353 "Attempting authentication using credentials from the properties");
4356 /* FIXME: If the url didn't contain username and password or we tried them
4357 * already, request a username and passwd from the application via some kind
4358 * of credentials request message */
4360 /* If we don't have a username and passwd at this point, bail out. */
4361 if (user == NULL || pass == NULL)
4364 /* Try to configure for each available authentication method, strongest to
4366 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
4367 /* Check if this method is available on the server */
4368 if ((method & avail_methods) == 0)
4371 /* Pass the credentials to the connection to try on the next request */
4372 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
4373 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
4374 * ignore it and end up retrying later */
4375 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
4376 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
4377 gst_rtsp_auth_method_to_string (method));
4382 if (method == GST_RTSP_AUTH_NONE)
4383 goto no_auth_available;
4389 /* Output an error indicating that we couldn't connect because there were
4390 * no supported authentication protocols */
4391 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4392 ("No supported authentication protocol was found"));
4397 /* We don't fire an error message, we just return FALSE and let the
4398 * normal NOT_AUTHORIZED error be propagated */
4403 static GstRTSPResult
4404 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4405 GstRTSPMessage * request, GstRTSPMessage * response,
4406 GstRTSPStatusCode * code)
4409 GstRTSPStatusCode thecode;
4410 gchar *content_base = NULL;
4414 if (!src->short_header)
4415 gst_rtsp_ext_list_before_send (src->extensions, request);
4417 GST_DEBUG_OBJECT (src, "sending message");
4420 gst_rtsp_message_dump (request);
4422 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
4426 gst_rtsp_connection_reset_timeout (conn);
4429 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
4434 gst_rtsp_message_dump (response);
4436 switch (response->type) {
4437 case GST_RTSP_MESSAGE_REQUEST:
4438 res = gst_rtspsrc_handle_request (src, conn, response);
4439 if (res == GST_RTSP_EEOF)
4442 goto handle_request_failed;
4444 case GST_RTSP_MESSAGE_RESPONSE:
4445 /* ok, a response is good */
4446 GST_DEBUG_OBJECT (src, "received response message");
4448 case GST_RTSP_MESSAGE_DATA:
4449 /* get next response */
4450 GST_DEBUG_OBJECT (src, "ignoring data response message");
4453 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4458 thecode = response->type_data.response.code;
4460 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
4462 /* if the caller wanted the result code, we store it. */
4466 /* If the request didn't succeed, bail out before doing any more */
4467 if (thecode != GST_RTSP_STS_OK)
4470 /* store new content base if any */
4471 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
4474 g_free (src->content_base);
4475 src->content_base = g_strdup (content_base);
4477 gst_rtsp_ext_list_after_send (src->extensions, request, response);
4484 gchar *str = gst_rtsp_strresult (res);
4486 if (res != GST_RTSP_EINTR) {
4487 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4488 ("Could not send message. (%s)", str));
4490 GST_WARNING_OBJECT (src, "send interrupted");
4499 GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
4502 /* if reconnect succeeds, try again */
4504 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
4508 /* only try once after reconnect, then fallthrough and error out */
4511 gchar *str = gst_rtsp_strresult (res);
4513 if (res != GST_RTSP_EINTR) {
4514 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4515 ("Could not receive message. (%s)", str));
4517 GST_WARNING_OBJECT (src, "receive interrupted");
4525 handle_request_failed:
4527 /* ERROR was posted */
4528 gst_rtsp_message_unset (response);
4533 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4534 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4535 ("The server closed the connection."));
4536 gst_rtsp_message_unset (response);
4543 * @src: the rtsp source
4544 * @conn: the connection to send on
4545 * @request: must point to a valid request
4546 * @response: must point to an empty #GstRTSPMessage
4547 * @code: an optional code result
4549 * send @request and retrieve the response in @response. optionally @code can be
4550 * non-NULL in which case it will contain the status code of the response.
4552 * If This function returns #GST_RTSP_OK, @response will contain a valid response
4553 * message that should be cleaned with gst_rtsp_message_unset() after usage.
4555 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
4556 * @response message) if the response code was not 200 (OK).
4558 * If the attempt results in an authentication failure, then this will attempt
4559 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
4562 * Returns: #GST_RTSP_OK if the processing was successful.
4564 static GstRTSPResult
4565 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4566 GstRTSPMessage * request, GstRTSPMessage * response,
4567 GstRTSPStatusCode * code)
4569 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
4570 GstRTSPResult res = GST_RTSP_ERROR;
4573 GstRTSPMethod method = GST_RTSP_INVALID;
4579 /* make sure we don't loop forever */
4583 /* save method so we can disable it when the server complains */
4584 method = request->type_data.request.method;
4587 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
4591 case GST_RTSP_STS_UNAUTHORIZED:
4592 if (gst_rtspsrc_setup_auth (src, response)) {
4593 /* Try the request/response again after configuring the auth info
4601 } while (retry == TRUE);
4603 /* If the user requested the code, let them handle errors, otherwise
4604 * post an error below */
4607 else if (int_code != GST_RTSP_STS_OK)
4608 goto error_response;
4615 GST_DEBUG_OBJECT (src, "got error %d", res);
4620 res = GST_RTSP_ERROR;
4622 switch (response->type_data.response.code) {
4623 case GST_RTSP_STS_NOT_FOUND:
4624 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
4625 response->type_data.response.reason));
4627 case GST_RTSP_STS_MOVED_PERMANENTLY:
4628 case GST_RTSP_STS_MOVE_TEMPORARILY:
4630 gchar *new_location;
4631 GstRTSPLowerTrans transports;
4633 GST_DEBUG_OBJECT (src, "got redirection");
4634 /* if we don't have a Location Header, we must error */
4635 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
4636 &new_location, 0) < 0)
4639 /* When we receive a redirect result, we go back to the INIT state after
4640 * parsing the new URI. The caller should do the needed steps to issue
4641 * a new setup when it detects this state change. */
4642 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
4644 /* save current transports */
4645 if (src->conninfo.url)
4646 transports = src->conninfo.url->transports;
4648 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
4650 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
4652 /* set old transports */
4653 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
4654 src->conninfo.url->transports = transports;
4656 src->need_redirect = TRUE;
4657 src->state = GST_RTSP_STATE_INIT;
4661 case GST_RTSP_STS_NOT_ACCEPTABLE:
4662 case GST_RTSP_STS_NOT_IMPLEMENTED:
4663 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
4664 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
4665 gst_rtsp_method_as_text (method));
4666 src->methods &= ~method;
4670 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4671 ("Got error response: %d (%s).", response->type_data.response.code,
4672 response->type_data.response.reason));
4675 /* if we return ERROR we should unset the response ourselves */
4676 if (res == GST_RTSP_ERROR)
4677 gst_rtsp_message_unset (response);
4683 static GstRTSPResult
4684 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
4685 GstRTSPMessage * response, GstRTSPSrc * src)
4687 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
4692 /* parse the response and collect all the supported methods. We need this
4693 * information so that we don't try to send an unsupported request to the
4697 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
4699 GstRTSPHeaderField field;
4705 /* reset supported methods */
4708 /* Try Allow Header first */
4709 field = GST_RTSP_HDR_ALLOW;
4712 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4713 if (indx == 0 && !respoptions) {
4714 /* if no Allow header was found then try the Public header... */
4715 field = GST_RTSP_HDR_PUBLIC;
4716 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4721 /* If we get here, the server gave a list of supported methods, parse
4722 * them here. The string is like:
4724 * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
4726 options = g_strsplit (respoptions, ",", 0);
4728 for (i = 0; options[i]; i++) {
4732 stripped = g_strstrip (options[i]);
4733 method = gst_rtsp_find_method (stripped);
4735 /* keep bitfield of supported methods */
4736 if (method != GST_RTSP_INVALID)
4737 src->methods |= method;
4739 g_strfreev (options);
4744 if (src->methods == 0) {
4745 /* neither Allow nor Public are required, assume the server supports
4746 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
4748 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
4749 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
4751 /* always assume PLAY, FIXME, extensions should be able to override
4753 src->methods |= GST_RTSP_PLAY;
4754 /* also assume it will support Range */
4755 src->seekable = TRUE;
4757 /* we need describe and setup */
4758 if (!(src->methods & GST_RTSP_DESCRIBE))
4760 if (!(src->methods & GST_RTSP_SETUP))
4768 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4769 ("Server does not support DESCRIBE."));
4774 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4775 ("Server does not support SETUP."));
4780 /* masks to be kept in sync with the hardcoded protocol order of preference
4782 static guint protocol_masks[] = {
4783 GST_RTSP_LOWER_TRANS_UDP,
4784 GST_RTSP_LOWER_TRANS_UDP_MCAST,
4785 GST_RTSP_LOWER_TRANS_TCP,
4789 static GstRTSPResult
4790 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
4791 GstRTSPLowerTrans protocols, gchar ** transports)
4795 gboolean add_udp_str;
4800 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
4805 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
4807 /* extension listed transports, use those */
4808 if (*transports != NULL)
4811 /* it's the default */
4812 add_udp_str = FALSE;
4814 /* the default RTSP transports */
4815 result = g_string_new ("");
4816 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
4817 GST_DEBUG_OBJECT (src, "adding UDP unicast");
4819 g_string_append (result, "RTP/AVP");
4821 g_string_append (result, "/UDP");
4822 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
4823 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
4824 GST_DEBUG_OBJECT (src, "adding UDP multicast");
4826 /* we don't have to allocate any UDP ports yet, if the selected transport
4827 * turns out to be multicast we can create them and join the multicast
4828 * group indicated in the transport reply */
4829 if (result->len > 0)
4830 g_string_append (result, ",");
4831 g_string_append (result, "RTP/AVP");
4833 g_string_append (result, "/UDP");
4834 g_string_append (result, ";multicast");
4835 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
4836 GST_DEBUG_OBJECT (src, "adding TCP");
4838 if (result->len > 0)
4839 g_string_append (result, ",");
4840 g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
4842 *transports = g_string_free (result, FALSE);
4844 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
4855 static GstRTSPResult
4856 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
4857 gint orig_rtpport, gint orig_rtcpport)
4860 gint nr_udp, nr_int;
4862 gint rtpport = 0, rtcpport = 0;
4865 src = stream->parent;
4867 /* find number of placeholders first */
4868 if (strstr (*transports, "%%i2"))
4870 else if (strstr (*transports, "%%i1"))
4875 if (strstr (*transports, "%%u2"))
4877 else if (strstr (*transports, "%%u1"))
4882 if (nr_udp == 0 && nr_int == 0)
4886 if (!orig_rtpport || !orig_rtcpport) {
4887 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
4890 rtpport = orig_rtpport;
4891 rtcpport = orig_rtcpport;
4895 str = g_string_new ("");
4897 while ((next = strstr (p, "%%"))) {
4898 g_string_append_len (str, p, next - p);
4899 if (next[2] == 'u') {
4901 g_string_append_printf (str, "%d", rtpport);
4902 else if (next[3] == '2')
4903 g_string_append_printf (str, "%d", rtcpport);
4905 if (next[2] == 'i') {
4907 g_string_append_printf (str, "%d", src->free_channel);
4908 else if (next[3] == '2')
4909 g_string_append_printf (str, "%d", src->free_channel + 1);
4914 /* append final part */
4915 g_string_append (str, p);
4917 g_free (*transports);
4918 *transports = g_string_free (str, FALSE);
4926 return GST_RTSP_ERROR;
4931 gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
4933 gboolean res = FALSE;
4937 const gchar *enc = NULL;
4939 s = gst_caps_get_structure (stream->caps, 0);
4940 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
4941 res = (strstr (enc, "-REAL") != NULL);
4947 /* Perform the SETUP request for all the streams.
4949 * We ask the server for a specific transport, which initially includes all the
4950 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
4951 * two local UDP ports that we send to the server.
4953 * Once the server replied with a transport, we configure the other streams
4954 * with the same transport.
4956 * This function will also configure the stream for the selected transport,
4957 * which basically means creating the pipeline.
4959 static GstRTSPResult
4960 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
4963 GstRTSPResult res = GST_RTSP_ERROR;
4964 GstRTSPMessage request = { 0 };
4965 GstRTSPMessage response = { 0 };
4966 GstRTSPStream *stream = NULL;
4967 GstRTSPLowerTrans protocols;
4968 GstRTSPStatusCode code;
4969 gboolean unsupported_real = FALSE;
4970 gint rtpport, rtcpport;
4974 if (src->conninfo.connection) {
4975 url = gst_rtsp_connection_get_url (src->conninfo.connection);
4976 /* we initially allow all configured lower transports. based on the URL
4977 * transports and the replies from the server we narrow them down. */
4978 protocols = url->transports & src->cur_protocols;
4981 protocols = src->cur_protocols;
4987 /* reset some state */
4988 src->free_channel = 0;
4989 src->interleaved = FALSE;
4990 src->need_activate = FALSE;
4991 /* keep track of next port number, 0 is random */
4992 src->next_port_num = src->client_port_range.min;
4993 rtpport = rtcpport = 0;
4995 if (G_UNLIKELY (src->streams == NULL))
4998 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4999 GstRTSPConnection *conn;
5004 stream = (GstRTSPStream *) walk->data;
5006 /* see if we need to configure this stream */
5007 if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
5008 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5010 stream->disabled = TRUE;
5014 /* merge/overwrite global caps */
5019 s = gst_caps_get_structure (stream->caps, 0);
5021 num = gst_structure_n_fields (src->props);
5022 for (j = 0; j < num; j++) {
5026 name = gst_structure_nth_field_name (src->props, j);
5027 val = gst_structure_get_value (src->props, name);
5028 gst_structure_set_value (s, name, val);
5030 GST_DEBUG_OBJECT (src, "copied %s", name);
5034 /* skip setup if we have no URL for it */
5035 if (stream->conninfo.location == NULL) {
5036 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5040 if (src->conninfo.connection == NULL) {
5041 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5042 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5045 conn = stream->conninfo.connection;
5047 conn = src->conninfo.connection;
5049 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5050 stream->conninfo.location);
5052 /* if we have a multicast connection, only suggest multicast from now on */
5053 if (stream->is_multicast)
5054 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5057 /* first selectable protocol */
5058 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5060 if (!protocol_masks[mask])
5064 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5065 protocol_masks[mask]);
5066 /* create a string with first transport in line */
5068 res = gst_rtspsrc_create_transports_string (src,
5069 protocols & protocol_masks[mask], &transports);
5070 if (res < 0 || transports == NULL)
5071 goto setup_transport_failed;
5073 if (strlen (transports) == 0) {
5074 g_free (transports);
5075 GST_DEBUG_OBJECT (src, "no transports found");
5080 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5082 /* replace placeholders with real values, this function will optionally
5083 * allocate UDP ports and other info needed to execute the setup request */
5084 res = gst_rtspsrc_prepare_transports (stream, &transports,
5085 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5087 g_free (transports);
5088 goto setup_transport_failed;
5091 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5093 /* create SETUP request */
5095 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5096 stream->conninfo.location);
5098 g_free (transports);
5099 goto create_request_failed;
5102 /* select transport, copy is made when adding to header so we can free it. */
5103 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5104 g_free (transports);
5106 /* if the user wants a non default RTP packet size we add the blocksize
5108 if (src->rtp_blocksize > 0) {
5109 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5110 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5115 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5118 /* handle the code ourselves */
5119 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5123 case GST_RTSP_STS_OK:
5125 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5126 gst_rtsp_message_unset (&request);
5127 gst_rtsp_message_unset (&response);
5128 /* cleanup of leftover transport */
5129 gst_rtspsrc_stream_free_udp (stream);
5130 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5131 * we might be in this case */
5132 if (stream->container && rtpport && rtcpport && !retry) {
5133 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5138 /* this transport did not go down well, but we may have others to try
5139 * that we did not send yet, try those and only give up then
5140 * but not without checking for lost cause/extension so we can
5141 * post a nicer/more useful error message later */
5142 if (!unsupported_real)
5143 unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
5144 /* select next available protocol, give up on this stream if none */
5146 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5148 if (!protocol_masks[mask] || unsupported_real)
5153 /* cleanup of leftover transport and move to the next stream */
5154 gst_rtspsrc_stream_free_udp (stream);
5155 goto response_error;
5158 /* parse response transport */
5160 gchar *resptrans = NULL;
5161 GstRTSPTransport transport = { 0 };
5163 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
5166 gst_rtspsrc_stream_free_udp (stream);
5170 /* parse transport, go to next stream on parse error */
5171 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
5172 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
5176 /* update allowed transports for other streams. once the transport of
5177 * one stream has been determined, we make sure that all other streams
5178 * are configured in the same way */
5179 switch (transport.lower_transport) {
5180 case GST_RTSP_LOWER_TRANS_TCP:
5181 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
5182 protocols = GST_RTSP_LOWER_TRANS_TCP;
5183 src->interleaved = TRUE;
5184 /* update free channels */
5186 MAX (transport.interleaved.min, src->free_channel);
5188 MAX (transport.interleaved.max, src->free_channel);
5189 src->free_channel++;
5191 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
5192 /* only allow multicast for other streams */
5193 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
5194 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
5196 case GST_RTSP_LOWER_TRANS_UDP:
5197 /* only allow unicast for other streams */
5198 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
5199 protocols = GST_RTSP_LOWER_TRANS_UDP;
5202 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
5203 transport.lower_transport);
5207 if (!stream->container || (!src->interleaved && !retry)) {
5208 /* now configure the stream with the selected transport */
5209 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
5210 GST_DEBUG_OBJECT (src,
5211 "could not configure stream %p transport, skipping stream",
5214 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
5215 /* retain the first allocated UDP port pair */
5216 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
5217 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
5220 /* we need to activate at least one streams when we detect activity */
5221 src->need_activate = TRUE;
5223 /* clean up our transport struct */
5224 gst_rtsp_transport_init (&transport);
5225 /* clean up used RTSP messages */
5226 gst_rtsp_message_unset (&request);
5227 gst_rtsp_message_unset (&response);
5231 /* store the transport protocol that was configured */
5232 src->cur_protocols = protocols;
5234 gst_rtsp_ext_list_stream_select (src->extensions, url);
5236 /* if there is nothing to activate, error out */
5237 if (!src->need_activate)
5238 goto nothing_to_activate;
5245 /* no transport possible, post an error and stop */
5246 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5247 ("Could not connect to server, no protocols left"));
5248 return GST_RTSP_ERROR;
5252 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5253 ("SDP contains no streams"));
5254 return GST_RTSP_ERROR;
5256 create_request_failed:
5258 gchar *str = gst_rtsp_strresult (res);
5260 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5261 ("Could not create request. (%s)", str));
5265 setup_transport_failed:
5267 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5268 ("Could not setup transport."));
5269 res = GST_RTSP_ERROR;
5274 const gchar *str = gst_rtsp_status_as_text (code);
5276 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5277 ("Error (%d): %s", code, GST_STR_NULL (str)));
5278 res = GST_RTSP_ERROR;
5283 gchar *str = gst_rtsp_strresult (res);
5285 if (res != GST_RTSP_EINTR) {
5286 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5287 ("Could not send message. (%s)", str));
5289 GST_WARNING_OBJECT (src, "send interrupted");
5296 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5297 ("Server did not select transport."));
5298 res = GST_RTSP_ERROR;
5301 nothing_to_activate:
5303 /* none of the available error codes is really right .. */
5304 if (unsupported_real) {
5305 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5306 (_("No supported stream was found. You might need to install a "
5307 "GStreamer RTSP extension plugin for Real media streams.")),
5310 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5311 (_("No supported stream was found. You might need to allow "
5312 "more transport protocols or may otherwise be missing "
5313 "the right GStreamer RTSP extension plugin.")), (NULL));
5315 return GST_RTSP_ERROR;
5319 gst_rtsp_message_unset (&request);
5320 gst_rtsp_message_unset (&response);
5326 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
5327 GstSegment * segment)
5330 GstRTSPTimeRange *therange;
5333 gst_rtsp_range_free (src->range);
5335 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
5336 GST_DEBUG_OBJECT (src, "parsed range %s", range);
5337 src->range = therange;
5339 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
5341 gst_segment_init (segment, GST_FORMAT_TIME);
5345 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
5346 therange->min.type, therange->min.seconds, therange->max.type,
5347 therange->max.seconds);
5349 if (therange->min.type == GST_RTSP_TIME_NOW)
5351 else if (therange->min.type == GST_RTSP_TIME_END)
5354 seconds = therange->min.seconds * GST_SECOND;
5356 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
5357 GST_TIME_ARGS (seconds));
5359 /* we need to start playback without clipping from the position reported by
5361 segment->start = seconds;
5362 segment->position = seconds;
5364 if (therange->max.type == GST_RTSP_TIME_NOW)
5366 else if (therange->max.type == GST_RTSP_TIME_END)
5369 seconds = therange->max.seconds * GST_SECOND;
5371 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
5372 GST_TIME_ARGS (seconds));
5374 /* live (WMS) server might send overflowed large max as its idea of infinity,
5375 * compensate to prevent problems later on */
5376 if (seconds != -1 && seconds < 0) {
5378 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
5381 /* live (WMS) might send min == max, which is not worth recording */
5382 if (segment->duration == -1 && seconds == segment->start)
5385 /* don't change duration with unknown value, we might have a valid value
5386 * there that we want to keep. */
5388 segment->duration = seconds;
5393 /* must be called with the RTSP state lock */
5394 static GstRTSPResult
5395 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
5401 /* prepare global stream caps properties */
5403 gst_structure_remove_all_fields (src->props);
5405 src->props = gst_structure_new_empty ("RTSPProperties");
5408 gst_sdp_message_dump (sdp);
5410 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
5412 gst_segment_init (&src->segment, GST_FORMAT_TIME);
5414 /* parse range for duration reporting. */
5419 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
5423 /* keep track of the range and configure it in the segment */
5424 if (gst_rtspsrc_parse_range (src, range, &src->segment))
5428 /* try to find a global control attribute. Note that a '*' means that we should
5429 * do aggregate control with the current url (so we don't do anything and
5430 * leave the current connection as is) */
5432 const gchar *control;
5435 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
5436 if (control == NULL)
5439 /* only take fully qualified urls */
5440 if (g_str_has_prefix (control, "rtsp://"))
5444 g_free (src->conninfo.location);
5445 src->conninfo.location = g_strdup (control);
5446 /* make a connection for this, if there was a connection already, nothing
5448 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
5449 GST_ERROR_OBJECT (src, "could not connect");
5452 /* we need to keep the control url separate from the connection url because
5453 * the rules for constructing the media control url need it */
5454 g_free (src->control);
5455 src->control = g_strdup (control);
5458 /* create streams */
5459 n_streams = gst_sdp_message_medias_len (sdp);
5460 for (i = 0; i < n_streams; i++) {
5461 gst_rtspsrc_create_stream (src, sdp, i);
5464 src->state = GST_RTSP_STATE_INIT;
5467 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
5470 /* reset our state */
5471 src->need_range = TRUE;
5474 src->state = GST_RTSP_STATE_READY;
5481 GST_ERROR_OBJECT (src, "setup failed");
5486 static GstRTSPResult
5487 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
5491 GstRTSPMessage request = { 0 };
5492 GstRTSPMessage response = { 0 };
5495 gchar *respcont = NULL;
5498 src->need_redirect = FALSE;
5500 /* can't continue without a valid url */
5501 if (G_UNLIKELY (src->conninfo.url == NULL)) {
5502 res = GST_RTSP_EINVAL;
5505 src->tried_url_auth = FALSE;
5507 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
5508 goto connect_failed;
5510 /* create OPTIONS */
5511 GST_DEBUG_OBJECT (src, "create options...");
5513 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
5514 src->conninfo.url_str);
5516 goto create_request_failed;
5519 GST_DEBUG_OBJECT (src, "send options...");
5522 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
5525 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5530 if (!gst_rtspsrc_parse_methods (src, &response))
5533 /* create DESCRIBE */
5534 GST_DEBUG_OBJECT (src, "create describe...");
5536 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
5537 src->conninfo.url_str);
5539 goto create_request_failed;
5541 /* we only accept SDP for now */
5542 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
5546 GST_DEBUG_OBJECT (src, "send describe...");
5549 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
5552 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5556 /* we only perform redirect for the describe, currently */
5557 if (src->need_redirect) {
5558 /* close connection, we don't have to send a TEARDOWN yet, ignore the
5560 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5562 gst_rtsp_message_unset (&request);
5563 gst_rtsp_message_unset (&response);
5569 /* it could be that the DESCRIBE method was not implemented */
5570 if (!src->methods & GST_RTSP_DESCRIBE)
5573 /* check if reply is SDP */
5574 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
5576 /* could not be set but since the request returned OK, we assume it
5577 * was SDP, else check it. */
5579 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
5580 goto wrong_content_type;
5583 /* get message body and parse as SDP */
5584 gst_rtsp_message_get_body (&response, &data, &size);
5585 if (data == NULL || size == 0)
5588 GST_DEBUG_OBJECT (src, "parse SDP...");
5589 gst_sdp_message_new (sdp);
5590 gst_sdp_message_parse_buffer (data, size, *sdp);
5592 /* clean up any messages */
5593 gst_rtsp_message_unset (&request);
5594 gst_rtsp_message_unset (&response);
5601 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
5602 ("No valid RTSP URL was provided"));
5607 gchar *str = gst_rtsp_strresult (res);
5609 if (res != GST_RTSP_EINTR) {
5610 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
5611 ("Failed to connect. (%s)", str));
5613 GST_WARNING_OBJECT (src, "connect interrupted");
5618 create_request_failed:
5620 gchar *str = gst_rtsp_strresult (res);
5622 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5623 ("Could not create request. (%s)", str));
5629 /* Don't post a message - the rtsp_send method will have
5630 * taken care of it because we passed NULL for the response code */
5635 /* error was posted */
5636 res = GST_RTSP_ERROR;
5641 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5642 ("Server does not support SDP, got %s.", respcont));
5643 res = GST_RTSP_ERROR;
5648 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5649 ("Server can not provide an SDP."));
5650 res = GST_RTSP_ERROR;
5655 if (src->conninfo.connection) {
5656 GST_DEBUG_OBJECT (src, "free connection");
5657 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5659 gst_rtsp_message_unset (&request);
5660 gst_rtsp_message_unset (&response);
5665 static GstRTSPResult
5666 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
5671 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
5673 if (src->sdp == NULL) {
5674 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
5678 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
5683 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
5690 GST_WARNING_OBJECT (src, "can't get sdp");
5691 src->open_error = TRUE;
5696 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
5697 src->open_error = TRUE;
5702 static GstRTSPResult
5703 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
5705 GstRTSPMessage request = { 0 };
5706 GstRTSPMessage response = { 0 };
5707 GstRTSPResult res = GST_RTSP_OK;
5711 GST_DEBUG_OBJECT (src, "TEARDOWN...");
5713 if (src->state < GST_RTSP_STATE_READY) {
5714 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
5721 /* construct a control url */
5723 control = src->control;
5725 control = src->conninfo.url_str;
5727 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
5730 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5731 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5733 GstRTSPConnInfo *info;
5735 /* try aggregate control first but do non-aggregate control otherwise */
5737 setup_url = control;
5738 else if ((setup_url = stream->conninfo.location) == NULL)
5741 if (src->conninfo.connection) {
5742 info = &src->conninfo;
5743 } else if (stream->conninfo.connection) {
5744 info = &stream->conninfo;
5748 if (!info->connected)
5753 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
5755 goto create_request_failed;
5758 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
5761 gst_rtspsrc_send (src, info->connection, &request, &response,
5765 /* FIXME, parse result? */
5766 gst_rtsp_message_unset (&request);
5767 gst_rtsp_message_unset (&response);
5770 /* early exit when we did aggregate control */
5776 /* close connections */
5777 GST_DEBUG_OBJECT (src, "closing connection...");
5778 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5779 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5780 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5781 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
5785 gst_rtspsrc_cleanup (src);
5787 src->state = GST_RTSP_STATE_INVALID;
5790 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
5795 create_request_failed:
5797 gchar *str = gst_rtsp_strresult (res);
5799 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5800 ("Could not create request. (%s)", str));
5806 gchar *str = gst_rtsp_strresult (res);
5808 gst_rtsp_message_unset (&request);
5809 if (res != GST_RTSP_EINTR) {
5810 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5811 ("Could not send message. (%s)", str));
5813 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
5820 GST_DEBUG_OBJECT (src,
5821 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
5826 /* RTP-Info is of the format:
5828 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
5830 * rtptime corresponds to the timestamp for the NPT time given in the header
5831 * seqbase corresponds to the next sequence number we received. This number
5832 * indicates the first seqnum after the seek and should be used to discard
5833 * packets that are from before the seek.
5836 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
5841 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
5843 infos = g_strsplit (rtpinfo, ",", 0);
5844 for (i = 0; infos[i]; i++) {
5846 GstRTSPStream *stream;
5850 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
5852 /* init values, types of seqbase and timebase are bigger than needed so we
5853 * can store -1 as uninitialized values */
5858 /* parse url, find stream for url.
5859 * parse seq and rtptime. The seq number should be configured in the rtp
5860 * depayloader or session manager to detect gaps. Same for the rtptime, it
5861 * should be used to create an initial time newsegment. */
5862 fields = g_strsplit (infos[i], ";", 0);
5863 for (j = 0; fields[j]; j++) {
5864 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
5865 /* remove leading whitespace */
5866 fields[j] = g_strchug (fields[j]);
5867 if (g_str_has_prefix (fields[j], "url=")) {
5868 /* get the url and the stream */
5870 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
5871 } else if (g_str_has_prefix (fields[j], "seq=")) {
5872 seqbase = atoi (fields[j] + 4);
5873 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
5874 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
5877 g_strfreev (fields);
5878 /* now we need to store the values for the caps of the stream */
5879 if (stream != NULL) {
5880 GST_DEBUG_OBJECT (src,
5881 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
5882 stream, seqbase, timebase);
5884 /* we have a stream, configure detected params */
5885 stream->seqbase = seqbase;
5886 stream->timebase = timebase;
5895 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
5900 interval = strtoul (rtcp, NULL, 10);
5901 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
5906 interval *= GST_MSECOND;
5908 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5909 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5911 /* already (optionally) retrieved this when configuring manager */
5912 if (stream->session) {
5913 GObject *rtpsession = stream->session;
5915 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
5917 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
5921 /* now it happens that (Xenon) server sending this may also provide bogus
5922 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
5923 * and just use RTP-Info to sync */
5925 GObjectClass *klass;
5927 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
5928 if (g_object_class_find_property (klass, "rtcp-sync")) {
5929 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
5930 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
5936 gst_rtspsrc_get_float (const gchar * dstr)
5938 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5940 /* canonicalise floating point string so we can handle float strings
5941 * in the form "24.930" or "24,930" irrespective of the current locale */
5942 g_strlcpy (s, dstr, sizeof (s));
5943 g_strdelimit (s, ",", '.');
5944 return g_ascii_strtod (s, NULL);
5948 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
5950 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5952 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
5953 g_strlcpy (val_str, "now", sizeof (val_str));
5955 if (segment->position == 0) {
5956 g_strlcpy (val_str, "0", sizeof (val_str));
5958 g_ascii_dtostr (val_str, sizeof (val_str),
5959 ((gdouble) segment->position) / GST_SECOND);
5962 return g_strdup_printf ("npt=%s-", val_str);
5966 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
5968 stream->timebase = -1;
5969 stream->seqbase = -1;
5973 stream->caps = gst_caps_make_writable (stream->caps);
5974 s = gst_caps_get_structure (stream->caps, 0);
5975 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
5979 static GstRTSPResult
5980 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
5982 GstRTSPResult res = GST_RTSP_OK;
5984 if (src->state < GST_RTSP_STATE_READY) {
5985 res = GST_RTSP_ERROR;
5986 if (src->open_error) {
5987 GST_DEBUG_OBJECT (src, "the stream was in error");
5991 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
5993 if ((res = gst_rtspsrc_open (src, async)) < 0) {
5994 GST_DEBUG_OBJECT (src, "failed to open stream");
6003 static GstRTSPResult
6004 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6006 GstRTSPMessage request = { 0 };
6007 GstRTSPMessage response = { 0 };
6008 GstRTSPResult res = GST_RTSP_OK;
6014 GST_DEBUG_OBJECT (src, "PLAY...");
6016 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6019 if (!(src->methods & GST_RTSP_PLAY))
6022 if (src->state == GST_RTSP_STATE_PLAYING)
6025 if (!src->conninfo.connection || !src->conninfo.connected)
6028 /* send some dummy packets before we activate the receive in the
6030 gst_rtspsrc_send_dummy_packets (src);
6032 /* activate receive elements;
6033 * only in async case, since receive elements may not have been affected
6034 * by overall state change (e.g. not around yet),
6035 * do not mess with state in sync case (e.g. seeking) */
6037 gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
6039 /* construct a control url */
6041 control = src->control;
6043 control = src->conninfo.url_str;
6045 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6046 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6048 GstRTSPConnection *conn;
6050 /* try aggregate control first but do non-aggregate control otherwise */
6052 setup_url = control;
6053 else if ((setup_url = stream->conninfo.location) == NULL)
6056 if (src->conninfo.connection) {
6057 conn = src->conninfo.connection;
6058 } else if (stream->conninfo.connection) {
6059 conn = stream->conninfo.connection;
6065 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
6067 goto create_request_failed;
6069 if (src->need_range) {
6070 hval = gen_range_header (src, segment);
6072 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
6076 if (segment->rate != 1.0) {
6077 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
6079 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
6081 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
6083 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
6087 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
6089 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6092 /* seek may have silently failed as it is not supported */
6093 if (!(src->methods & GST_RTSP_PLAY)) {
6094 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
6095 /* obviously it is supported as we made it here */
6096 src->methods |= GST_RTSP_PLAY;
6097 src->seekable = FALSE;
6098 /* but there is nothing to parse in the response,
6099 * so convey we have no idea and not to expect anything particular */
6100 clear_rtp_base (src, stream);
6104 /* need to do for all streams */
6105 for (run = src->streams; run; run = g_list_next (run))
6106 clear_rtp_base (src, (GstRTSPStream *) run->data);
6108 /* NOTE the above also disables npt based eos detection */
6109 /* and below forces position to 0,
6110 * which is visible feedback we lost the plot */
6111 segment->start = segment->position = src->last_pos;
6114 gst_rtsp_message_unset (&request);
6116 /* parse RTP npt field. This is the current position in the stream (Normal
6117 * Play Time) and should be put in the NEWSEGMENT position field. */
6118 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
6120 gst_rtspsrc_parse_range (src, hval, segment);
6122 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
6123 segment->rate = 1.0;
6125 /* parse Speed header. This is the intended playback rate of the stream
6126 * and should be put in the NEWSEGMENT rate field. */
6127 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
6128 0) == GST_RTSP_OK) {
6129 segment->rate = gst_rtspsrc_get_float (hval);
6130 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
6131 &hval, 0) == GST_RTSP_OK) {
6132 segment->rate = gst_rtspsrc_get_float (hval);
6135 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
6136 * for the RTP packets. If this is not present, we assume all starts from 0...
6137 * This is info for the RTP session manager that we pass to it in caps. */
6139 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
6140 &hval, hval_idx++) == GST_RTSP_OK)
6141 gst_rtspsrc_parse_rtpinfo (src, hval);
6143 /* some servers indicate RTCP parameters in PLAY response,
6144 * rather than properly in SDP */
6145 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
6146 &hval, 0) == GST_RTSP_OK)
6147 gst_rtspsrc_handle_rtcp_interval (src, hval);
6149 gst_rtsp_message_unset (&response);
6151 /* early exit when we did aggregate control */
6155 /* set again when needed */
6156 src->need_range = FALSE;
6158 /* configure the caps of the streams after we parsed all headers. */
6159 gst_rtspsrc_configure_caps (src, segment);
6161 src->running = TRUE;
6162 src->base_time = -1;
6163 src->state = GST_RTSP_STATE_PLAYING;
6166 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
6167 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6168 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6169 stream->discont = TRUE;
6174 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
6181 GST_DEBUG_OBJECT (src, "failed to open stream");
6186 GST_DEBUG_OBJECT (src, "PLAY is not supported");
6191 GST_DEBUG_OBJECT (src, "we were already PLAYING");
6194 create_request_failed:
6196 gchar *str = gst_rtsp_strresult (res);
6198 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6199 ("Could not create request. (%s)", str));
6205 gchar *str = gst_rtsp_strresult (res);
6207 gst_rtsp_message_unset (&request);
6208 if (res != GST_RTSP_EINTR) {
6209 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6210 ("Could not send message. (%s)", str));
6212 GST_WARNING_OBJECT (src, "PLAY interrupted");
6219 static GstRTSPResult
6220 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
6222 GstRTSPResult res = GST_RTSP_OK;
6223 GstRTSPMessage request = { 0 };
6224 GstRTSPMessage response = { 0 };
6228 GST_DEBUG_OBJECT (src, "PAUSE...");
6230 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6233 if (!(src->methods & GST_RTSP_PAUSE))
6236 if (src->state == GST_RTSP_STATE_READY)
6239 if (!src->conninfo.connection || !src->conninfo.connected)
6242 /* construct a control url */
6244 control = src->control;
6246 control = src->conninfo.url_str;
6248 /* loop over the streams. We might exit the loop early when we could do an
6249 * aggregate control */
6250 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6251 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6252 GstRTSPConnection *conn;
6255 /* try aggregate control first but do non-aggregate control otherwise */
6257 setup_url = control;
6258 else if ((setup_url = stream->conninfo.location) == NULL)
6261 if (src->conninfo.connection) {
6262 conn = src->conninfo.connection;
6263 } else if (stream->conninfo.connection) {
6264 conn = stream->conninfo.connection;
6270 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
6271 ("Sending PAUSE request"));
6274 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
6276 goto create_request_failed;
6278 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6281 gst_rtsp_message_unset (&request);
6282 gst_rtsp_message_unset (&response);
6284 /* exit early when we did agregate control */
6290 src->state = GST_RTSP_STATE_READY;
6294 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
6301 GST_DEBUG_OBJECT (src, "failed to open stream");
6306 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
6311 GST_DEBUG_OBJECT (src, "we were already PAUSED");
6314 create_request_failed:
6316 gchar *str = gst_rtsp_strresult (res);
6318 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6319 ("Could not create request. (%s)", str));
6325 gchar *str = gst_rtsp_strresult (res);
6327 gst_rtsp_message_unset (&request);
6328 if (res != GST_RTSP_EINTR) {
6329 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6330 ("Could not send message. (%s)", str));
6332 GST_WARNING_OBJECT (src, "PAUSE interrupted");
6340 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
6342 GstRTSPSrc *rtspsrc;
6344 rtspsrc = GST_RTSPSRC (bin);
6346 switch (GST_MESSAGE_TYPE (message)) {
6347 case GST_MESSAGE_EOS:
6348 gst_message_unref (message);
6350 case GST_MESSAGE_ELEMENT:
6352 const GstStructure *s = gst_message_get_structure (message);
6354 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
6355 gboolean ignore_timeout;
6357 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
6359 GST_OBJECT_LOCK (rtspsrc);
6360 ignore_timeout = rtspsrc->ignore_timeout;
6361 rtspsrc->ignore_timeout = TRUE;
6362 GST_OBJECT_UNLOCK (rtspsrc);
6364 /* we only act on the first udp timeout message, others are irrelevant
6365 * and can be ignored. */
6366 if (!ignore_timeout)
6367 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
6369 gst_message_unref (message);
6372 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6375 case GST_MESSAGE_ERROR:
6378 GstRTSPStream *stream;
6381 udpsrc = GST_MESSAGE_SRC (message);
6383 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
6384 GST_ELEMENT_NAME (udpsrc));
6386 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
6390 /* we ignore the RTCP udpsrc */
6391 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
6394 /* if we get error messages from the udp sources, that's not a problem as
6395 * long as not all of them error out. We also don't really know what the
6396 * problem is, the message does not give enough detail... */
6397 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
6398 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
6399 if (ret != GST_FLOW_OK)
6403 gst_message_unref (message);
6407 /* fatal but not our message, forward */
6408 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6413 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6419 /* the thread where everything happens */
6421 gst_rtspsrc_thread (GstRTSPSrc * src)
6425 gboolean running = FALSE;
6427 GST_OBJECT_LOCK (src);
6428 cmd = src->loop_cmd;
6429 src->loop_cmd = CMD_WAIT;
6430 GST_DEBUG_OBJECT (src, "got command %d", cmd);
6432 /* we got the message command, so ensure communication is possible again */
6433 gst_rtspsrc_connection_flush (src, FALSE);
6435 /* we allow these to be interrupted */
6436 if (cmd == CMD_LOOP || cmd == CMD_CLOSE || cmd == CMD_PAUSE)
6437 src->waiting = TRUE;
6438 GST_OBJECT_UNLOCK (src);
6442 ret = gst_rtspsrc_open (src, TRUE);
6445 ret = gst_rtspsrc_play (src, &src->segment, TRUE);
6446 if (ret == GST_RTSP_OK)
6450 ret = gst_rtspsrc_pause (src, TRUE, TRUE);
6451 if (ret == GST_RTSP_OK)
6455 ret = gst_rtspsrc_close (src, TRUE, FALSE);
6458 running = gst_rtspsrc_loop (src);
6461 ret = gst_rtspsrc_reconnect (src, FALSE);
6462 if (ret == GST_RTSP_OK)
6469 GST_OBJECT_LOCK (src);
6470 /* and go back to sleep */
6471 if (src->loop_cmd == CMD_WAIT) {
6473 src->loop_cmd = CMD_LOOP;
6475 gst_task_pause (src->task);
6478 src->waiting = FALSE;
6479 GST_OBJECT_UNLOCK (src);
6483 gst_rtspsrc_start (GstRTSPSrc * src)
6485 GST_DEBUG_OBJECT (src, "starting");
6487 GST_OBJECT_LOCK (src);
6489 src->loop_cmd = CMD_WAIT;
6491 if (src->task == NULL) {
6492 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src);
6493 if (src->task == NULL)
6496 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
6498 GST_OBJECT_UNLOCK (src);
6505 GST_ERROR_OBJECT (src, "failed to create task");
6511 gst_rtspsrc_stop (GstRTSPSrc * src)
6515 GST_DEBUG_OBJECT (src, "stopping");
6517 /* also cancels pending task */
6518 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
6520 GST_OBJECT_LOCK (src);
6521 if ((task = src->task)) {
6523 GST_OBJECT_UNLOCK (src);
6525 gst_task_stop (task);
6527 /* make sure it is not running */
6528 GST_RTSP_STREAM_LOCK (src);
6529 GST_RTSP_STREAM_UNLOCK (src);
6531 /* now wait for the task to finish */
6532 gst_task_join (task);
6534 /* and free the task */
6535 gst_object_unref (GST_OBJECT (task));
6537 GST_OBJECT_LOCK (src);
6539 GST_OBJECT_UNLOCK (src);
6541 /* ensure synchronously all is closed and clean */
6542 gst_rtspsrc_close (src, FALSE, TRUE);
6547 static GstStateChangeReturn
6548 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
6550 GstRTSPSrc *rtspsrc;
6551 GstStateChangeReturn ret;
6553 rtspsrc = GST_RTSPSRC (element);
6555 switch (transition) {
6556 case GST_STATE_CHANGE_NULL_TO_READY:
6557 if (!gst_rtspsrc_start (rtspsrc))
6560 case GST_STATE_CHANGE_READY_TO_PAUSED:
6561 /* init some state */
6562 rtspsrc->cur_protocols = rtspsrc->protocols;
6563 /* first attempt, don't ignore timeouts */
6564 rtspsrc->ignore_timeout = FALSE;
6565 rtspsrc->open_error = FALSE;
6566 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN);
6568 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6569 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6570 /* unblock the tcp tasks and make the loop waiting */
6571 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT);
6573 case GST_STATE_CHANGE_PAUSED_TO_READY:
6579 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
6580 if (ret == GST_STATE_CHANGE_FAILURE)
6583 switch (transition) {
6584 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6585 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY);
6587 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6588 /* send pause request and keep the idle task around */
6589 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE);
6590 ret = GST_STATE_CHANGE_NO_PREROLL;
6592 case GST_STATE_CHANGE_READY_TO_PAUSED:
6593 ret = GST_STATE_CHANGE_NO_PREROLL;
6595 case GST_STATE_CHANGE_PAUSED_TO_READY:
6596 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE);
6598 case GST_STATE_CHANGE_READY_TO_NULL:
6599 gst_rtspsrc_stop (rtspsrc);
6610 GST_DEBUG_OBJECT (rtspsrc, "start failed");
6611 return GST_STATE_CHANGE_FAILURE;
6616 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
6619 GstRTSPSrc *rtspsrc;
6621 rtspsrc = GST_RTSPSRC (element);
6623 if (GST_EVENT_IS_DOWNSTREAM (event)) {
6624 res = gst_rtspsrc_push_event (rtspsrc, event, TRUE);
6626 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
6633 /*** GSTURIHANDLER INTERFACE *************************************************/
6636 gst_rtspsrc_uri_get_type (GType type)
6641 static const gchar *const *
6642 gst_rtspsrc_uri_get_protocols (GType type)
6644 static const gchar *protocols[] =
6645 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", NULL };
6651 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
6653 GstRTSPSrc *src = GST_RTSPSRC (handler);
6655 /* FIXME: make thread-safe */
6656 return g_strdup (src->conninfo.location);
6660 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
6665 GstRTSPUrl *newurl = NULL;
6666 GstSDPMessage *sdp = NULL;
6668 src = GST_RTSPSRC (handler);
6670 /* same URI, we're fine */
6671 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
6674 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
6675 if ((res = gst_sdp_message_new (&sdp) < 0))
6678 GST_DEBUG_OBJECT (src, "parsing SDP message");
6679 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
6683 GST_DEBUG_OBJECT (src, "parsing URI");
6684 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
6688 /* if worked, free previous and store new url object along with the original
6690 GST_DEBUG_OBJECT (src, "configuring URI");
6691 g_free (src->conninfo.location);
6692 src->conninfo.location = g_strdup (uri);
6693 gst_rtsp_url_free (src->conninfo.url);
6694 src->conninfo.url = newurl;
6695 g_free (src->conninfo.url_str);
6697 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
6699 src->conninfo.url_str = NULL;
6702 gst_sdp_message_free (src->sdp);
6704 src->from_sdp = sdp != NULL;
6706 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
6707 GST_DEBUG_OBJECT (src, "request uri is: %s",
6708 GST_STR_NULL (src->conninfo.url_str));
6715 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
6720 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
6721 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6722 "Could not create SDP");
6727 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
6728 GST_STR_NULL (uri));
6729 gst_sdp_message_free (sdp);
6730 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6736 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
6737 GST_STR_NULL (uri), res);
6738 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6739 "Invalid RTSP URI");
6745 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
6747 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
6749 iface->get_type = gst_rtspsrc_uri_get_type;
6750 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
6751 iface->get_uri = gst_rtspsrc_uri_get_uri;
6752 iface->set_uri = gst_rtspsrc_uri_set_uri;